| 00:00.30 | J4k3 | theres engliush and english |
| 00:00.39 | drmessano | and engrish |
| 00:00.40 | obnauticus | and engrish |
| 00:00.41 | obnauticus | LOL |
| 00:00.46 | J4k3 | cuz for some reason those people that hump the queen also like to throw random extra letters in |
| 00:00.48 | drmessano | lol |
| 00:00.51 | J4k3 | especially u's |
| 00:00.58 | drmessano | colour |
| 00:01.01 | drmessano | favour |
| 00:01.02 | J4k3 | and y's |
| 00:01.04 | obnauticus | i seriously said that at the same exact time i recieved it drmessano |
| 00:01.05 | J4k3 | tyre |
| 00:01.13 | drmessano | lol |
| 00:01.30 | drmessano | fyre |
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| 00:04.28 | *** join/#asterisk jmacz (n=jmacz@190.90.35.212) |
| 00:05.54 | eric2 | JT: my phone is connected to a linksys wrt54xx router |
| 00:08.27 | *** join/#asterisk phillipk (n=phillipk@fw.datafax.net) |
| 00:15.41 | JT | extra letters, ie. correct english, and they weren't added so much as US variants of english arbitarily chopped letters and bastardised english |
| 00:15.50 | JT | eric2: ok, and what are you trying to call? |
| 00:16.28 | drmessano | I like to think we made it more efficient |
| 00:18.49 | eric2 | today I was on hold with a 1-900 number, when she came to speak, I could hear her, but she couldn't hear me |
| 00:19.25 | eric2 | err, 1-800 I mean |
| 00:21.27 | Qwell | eric2: suuuuuuuuure you do :) |
| 00:21.39 | drmessano | LOL |
| 00:21.41 | drmessano | Dude |
| 00:21.53 | drmessano | You called a 900 number and she put you on hold |
| 00:23.33 | eric2 | haha, I knew you'd get a kick out of that.... |
| 00:23.50 | drmessano | Naah.. Just feelings of pity :( |
| 00:24.07 | eric2 | no, I called a 1-800 #, was on hold for 20 minutes.. when she came to talk, she couldn't hear me |
| 00:24.21 | eric2 | and its happened once or twice when dialing a local number too |
| 00:24.38 | JT | eric2: ok, i'm still having difficulty getting this info |
| 00:24.45 | JT | eric2: HOW are you making PSTN calls? |
| 00:24.53 | eric2 | virtual pri |
| 00:25.01 | JT | i see |
| 00:25.03 | eric2 | sip termination I guess |
| 00:25.05 | JT | so SIP |
| 00:25.09 | eric2 | yes |
| 00:25.17 | JT | make sure canreinvite=no |
| 00:25.21 | eric2 | k |
| 00:25.21 | *** join/#asterisk cesar_CR (n=cesar@celord.ice.co.cr) |
| 00:25.21 | JT | and follow the other stuff in |
| 00:25.23 | JT | ~sipnat |
| 00:25.24 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 00:25.28 | JT | first url |
| 00:25.32 | eric2 | k, tx |
| 00:26.17 | eric2 | ok, canreinvite is set to 'no' |
| 00:26.41 | eric2 | on to the reading |
| 00:28.05 | drmessano | Qwell: You there? |
| 00:29.42 | Qwell | ? |
| 00:29.59 | drmessano | I got my $7 (lol) Bluetooth dongles from china... I plugged them into XP to see if it would autodetect it (as my other dongle did), but it seems like it needs this BlueSoleil software installed first.. Does that tell you anything about the chipset or how difficult setting it up in Linux will be? |
| 00:30.27 | Qwell | no idea |
| 00:30.34 | drmessano | Aight.. it was worth a shot |
| 00:30.36 | Qwell | I think the software you installed was driver+malware |
| 00:30.45 | drmessano | LOL |
| 00:30.49 | drmessano | I havent installed anything yet |
| 00:31.23 | drmessano | I always investigate the drivers and shit first when I get something that Windows SHOULD detect, but doesnt |
| 00:31.36 | drmessano | ..for that very reason |
| 00:31.46 | drmessano | Free Subscription to ChineseMonkey.com or something |
| 00:32.01 | Qwell | drmessano: I have an extra invite if you want it |
| 00:32.18 | drmessano | I got 3 dongles, so i'll have 2 extra :) |
| 00:36.59 | drmessano | HA awesome |
| 00:37.17 | drmessano | I googled the goofy device name that XP presents me with |
| 00:37.26 | drmessano | ISSCEDRBTA |
| 00:37.46 | drmessano | and someone updated the inf in XP to detect the hardware ID... |
| 00:37.52 | drmessano | So no goofy driver needed |
| 00:38.04 | drmessano | Guess I will try it on Linux now.. |
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| 00:42.03 | vap0rtranz | i'm failing on the telco lingo for "give me a line and let me dial something myself". that simply done by dial() options? |
| 00:42.36 | drmessano | Ok |
| 00:42.37 | riddlebox | can someone put me on the right path, I want to show when a line is in use? |
| 00:42.44 | drmessano | So the BT dongle is working in Asterisk too |
| 00:42.45 | drmessano | $7 |
| 00:44.27 | vap0rtranz | riddlebox: channelvariables.txt ? |
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| 00:52.53 | drmessano | Asterisk is a tornado repellent |
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| 00:54.49 | rickross | how does one set up "estension-style" dialing of a MeetMe conference number from an inbound IVR menu? |
| 00:56.48 | drmessano | rickross: Is this for FreePBX? |
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| 00:57.43 | rickross | drmessano: I am using freePBX 2.4.0, yeas |
| 00:57.50 | rickross | andasterisk 1.6.0b4 |
| 00:58.05 | drmessano | Youre asking in the wrong place |
| 00:58.14 | drmessano | FreePBX isn't supported here |
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| 00:58.30 | rickross | MeetMe is asterisk, and the IVR is , too - isn't it? |
| 00:58.53 | drmessano | FreePBX isn't supported here |
| 00:59.08 | rickross | afaik, the FreePBX part is actually incidental |
| 00:59.20 | drmessano | No, it's not |
| 00:59.32 | rickross | and the issue is that the MeetMe conference number doesn't get treated as an extension |
| 00:59.41 | drmessano | Which is a FreePBX issue |
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| 01:02.03 | fujin | outtolunc: fwiw, here's a perl snippet to convert sip.conf for sippeers realtime. http://rafb.net/p/O9uRQm11.html |
| 01:02.04 | drmessano | I was trying to guide him to the other channel when I could help him.. I got his attention now.. lol |
| 01:02.29 | fujin | ahcrap |
| 01:02.31 | fujin | had a u/p in it |
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| 01:02.34 | fujin | rofl |
| 01:02.41 | fujin | wtb: coffee |
| 01:02.45 | outtolunc | thanks but i don't need one <G> |
| 01:02.50 | outtolunc | i'll put it on the site tho |
| 01:02.56 | fujin | please don't, not that one |
| 01:03.01 | outtolunc | k |
| 01:03.25 | fujin | (don't even try guys, it's locked down by iptables and mysql authentication) |
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| 01:09.24 | Yourname`` | goatse |
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| 01:22.56 | drmessano | http://www.theregister.co.uk/2008/03/04/gary_gygax_dies_at_69/ <--- Missed his saving throw |
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| 01:29.31 | Mavvie | drmessano: I'll get his boots! And his sword! |
| 01:29.39 | SteveTotaro | man is getting an ati radeon 9000 setup for xorg difficult! |
| 01:29.47 | Qwell | SteveTotaro: nvidia |
| 01:30.01 | SteveTotaro | i can only get it working with vesa driver |
| 01:30.01 | Qwell | ATI + Linux is pretty terrible...still |
| 01:30.17 | SteveTotaro | it's ok but dvd playback sucks |
| 01:31.04 | SteveTotaro | yeah, i guess i will have to pilfer an nvidia |
| 01:31.47 | SteveTotaro | i figured i could hack my way around it somehow but too frustrating for stinking dvds |
| 01:33.59 | SteveTotaro | i am watching the 4400 series, never saw it but then what is really the point since it is canceled? still pretty decent i guess. |
| 01:34.30 | Yourname`` | app_queue by who? |
| 01:37.33 | SteveTotaro | is k9copy done yet? |
| 01:37.58 | drmessano | lol Mavvie |
| 01:38.20 | SteveTotaro | i have to get these netflix back to the post office really soon |
| 01:38.42 | drmessano | Netflix is too inconvenient |
| 01:38.56 | SteveTotaro | netflix rules |
| 01:39.00 | drmessano | All that walking to the mailbox.. they can kiss my ass |
| 01:39.04 | SteveTotaro | i have had them since day one |
| 01:39.11 | drmessano | I exercise for NO ONE |
| 01:39.19 | SteveTotaro | my girl brings in the mail and takes out the trash |
| 01:40.00 | drmessano | Lucky.. not many guys have netflix fetchchix |
| 01:40.29 | SteveTotaro | i just have to make sure my pit bull isn't outside when the mailman comes |
| 01:40.46 | drmessano | They have spray for that :) |
| 01:41.05 | SteveTotaro | don't think it would do anything but piss him off more |
| 01:41.40 | SteveTotaro | he fell out of my Cadillac Eldorado going 40MPH at least |
| 01:41.48 | drmessano | lol |
| 01:41.54 | SteveTotaro | right out the driver's side window |
| 01:42.05 | Yourname`` | I have this anonymous company that hand delivers DVDs to my house. And comes and picks it up from me when I call them "It's in my postbox." |
| 01:42.24 | SteveTotaro | onto the pavement, i was shaking and freaked out |
| 01:42.40 | SteveTotaro | took him to the emergency vet, nothing but road rash |
| 01:42.46 | drmessano | Damn |
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| 01:43.15 | SteveTotaro | he is 85 pounds of rippling muscle |
| 01:43.29 | SteveTotaro | can hang by his teeth in mid air for twenty minutes |
| 01:43.54 | SteveTotaro | well not so long anymore, he is 12 years old now but still a puppy |
| 01:46.05 | SteveTotaro | http://www.kodakgallery.com/Slideshow.jsp?Uc=chh334e.8w43mw3q&Uy=yh22z&Upost_signin=Slideshow.jsp%3Fmode%3Dfromshare&Ux=0&mode=fromshare&conn_speed=1 |
| 01:46.13 | drmessano | Ron Paul is gonna win |
| 01:46.25 | SteveTotaro | he won my heart |
| 01:46.41 | J4k3 | nigga, you gay. |
| 01:47.25 | SteveTotaro | ~ronpaul |
| 01:47.30 | SteveTotaro | ~ron paul |
| 01:47.31 | jbot | Ron paul could kick chuck Noris; Arse |
| 01:47.35 | SteveTotaro | ~gay |
| 01:47.36 | jbot | No, YOU'RE gay. |
| 01:48.35 | SteveTotaro | anyone that wants to get rid of the the fed and federal income tax has my heart |
| 01:48.39 | [TK]D-Fender | drmessano : Sure he'll win... its just he's not playing the "Get elected president of the USA" game. |
| 01:48.46 | drmessano | HAH |
| 01:48.55 | drmessano | He's winning Digg |
| 01:49.24 | [TK]D-Fender | drmessano, I hate to say it, but it really looks like Barack has a SERIOUS chance of winning this solid. He appeals far better to the masses. |
| 01:50.05 | [TK]D-Fender | drmessano, Actually.... RP's Diggs have slowed to an absolute crawl. its been nothing but Barack and Hillary all over. McCain hardly even registers online. |
| 01:50.13 | drmessano | Hillary isn't going down without a fight.. She's gonna pull the "Lewinsky Maneuver" and shock them all |
| 01:50.56 | drmessano | Yeah.. the RP stuff has calmed down.. but it was fanboyville for some time |
| 01:51.02 | [TK]D-Fender | drmessano, wouldn't put money on her. |
| 01:51.31 | drmessano | At this point, I won't either.. She's cracking |
| 01:51.37 | [TK]D-Fender | drmessano, Oh yes, 2-3 months ago it was all RP fever... however he hasn't advertised worth shit and made his presence credible. |
| 01:52.22 | drmessano | Jon Stewart said it best.. "So, which democrat is everyone voting for?" |
| 01:52.23 | J4k3 | I decided to not vote |
| 01:52.45 | J4k3 | the choices for the local elections were pretty slim, and the choices for presidential wasn't worth the effort. |
| 01:52.54 | J4k3 | shitchoice vs shitchoice vs republicans |
| 01:52.59 | drmessano | Local elections make it worth it |
| 01:53.06 | drmessano | Vote for Bucky |
| 01:53.19 | [TK]D-Fender | drmessano, Kucinich is out :( RP is in the wrong sport let alone ball-park, Hillary is tanking :), an Barack running strong :} |
| 01:53.42 | drmessano | "I'm Shail Failover, and I approve this message" <-- Local rules |
| 01:53.43 | [TK]D-Fender | drmessano, So far as long as no republican save RP makes it I'm happy enough. |
| 01:53.47 | J4k3 | well, for most things here its just 1 repub/1 democrat... the only one with any hype going on is the republican candidates for county commissioner, and personally I hope they fuck up this time around royally so I can get the job next election |
| 01:54.17 | J4k3 | ron paul isn't going to win anything |
| 01:54.24 | J4k3 | its mccain or huckleberry |
| 01:54.36 | drmessano | J4k3: He could still win Reddit |
| 01:54.36 | J4k3 | and it'll likely be mccain, due to idiot democrat crossover voters. |
| 01:54.53 | drmessano | Friendster is undecided |
| 01:55.28 | [TK]D-Fender | J4k3, Again, I put no odds on RP at this point. Huckabee is thankfully fading. While McCain is a militant, Huckabee is a friggen CRUSADER. |
| 01:55.41 | J4k3 | mccain is an easy republican loss |
| 01:55.45 | drmessano | If McCain wins, i'm gonna assassinate Lincoln again |
| 01:55.47 | J4k3 | but... hillary vs obama |
| 01:55.57 | J4k3 | ballsy bitch that nobody likes vs black man with no backbone. |
| 01:56.29 | [TK]D-Fender | J4k3, Hillary is possibly worse than Mccain. |
| 01:57.14 | drmessano | At least Obama won't send us to war against any more muslims.. :D |
| 01:59.04 | [TK]D-Fender | drmessano, yup. I can be happy with his winning. Not I guy I see screwing up the county much. |
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| 02:11.42 | [TK]D-Fender | *b00m* |
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| 02:12.36 | drmessano | SteveTotaro |
| 02:12.36 | SteveTotaro | yo |
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| 02:12.50 | SteveTotaro | my HTPC is acting up |
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| 02:13.04 | SteveTotaro | pidgin keeps crashing |
| 02:13.04 | drmessano | In setting up chan+mobile, I got this string for making my box discoverable.. |
| 02:13.04 | drmessano | dbus-send --system --type=method_call --print-reply --dest=org.bluez /org/bluez/hci0 org.bluez.Adapter.SetMode string:discoverable |
| 02:13.55 | drmessano | I was thinking about creating an extension to execute that with System(), but not sure I could wrap that in anything to make it work.. |
| 02:13.56 | SteveTotaro | i did that somewhere in a bluez conf file |
| 02:14.06 | SteveTotaro | not that string but set it to be discoverable |
| 02:14.32 | drmessano | ah |
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| 02:14.33 | drmessano | Just leaving discovery on |
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| 02:14.33 | SteveTotaro | yeah, pin is required to pair |
| 02:14.55 | drmessano | Yeah |
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| 02:14.56 | SteveTotaro | my stuff is locked in a trusted data center |
| 02:15.19 | SteveTotaro | basically in a bunker |
| 02:15.39 | SteveTotaro | disguised as a farm |
| 02:16.06 | JT | lol |
| 02:16.18 | JT | in other words you have a pc in a barn on a dsl line |
| 02:16.27 | SteveTotaro | nope |
| 02:16.29 | drmessano | cockadoodledoo |
| 02:16.38 | SteveTotaro | it is a POP room for sure |
| 02:18.05 | SteveTotaro | whole b class subnet |
| 02:18.51 | SteveTotaro | ~class |
| 02:18.52 | jbot | extra, extra, read all about it, class is over |
| 02:19.08 | JT | and it looks like a farm? |
| 02:19.10 | SteveTotaro | just making sure i wasn't bumped again |
| 02:19.15 | SteveTotaro | it is a farm |
| 02:19.23 | SteveTotaro | or was |
| 02:19.27 | JT | heh |
| 02:19.32 | SteveTotaro | it qualifies as a farm |
| 02:19.41 | SteveTotaro | tax free gas |
| 02:20.36 | SteveTotaro | only thing that grows there is grass and LOTS of deer |
| 02:22.53 | SteveTotaro | http://www.schoffstall.com/the_farm.htm |
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| 02:25.05 | J4k3 | he |
| 02:25.18 | SteveTotaro | ~Kumbang |
| 02:25.26 | J4k3 | so do they run a police force to keep the mexicans with backhoes away from their rural-ass fiber thats prolly buried a whole 4-6 inches below the dirt. |
| 02:25.29 | J4k3 | ;) |
| 02:27.51 | plik | hah! http://www.flickr.com/photos/billadams/321845104/ |
| 02:28.20 | JT | that page doesn't say anything about a datacentre? |
| 02:28.46 | J4k3 | but goddamn if I can get access to any fiber, much less fiber thats got any redundancy, or even properly buried. |
| 02:30.47 | Yourname`` | Is AgentLogin() being removed? |
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| 02:32.18 | dlynes | Anyone on here offer reasonable quality toll-free dids? |
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| 02:33.29 | dlynes | I'm looking for a toll-free did that is accessible from anywhere in the continental US, and Canada |
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| 02:34.05 | SteveTotaro | quick, download them while you can |
| 02:34.07 | SteveTotaro | http://www.sedition.org.au/CAL207_V10.5.pdf |
| 02:34.11 | SteveTotaro | http://www.sedition.org.au/calypso_c035.pdf |
| 02:34.13 | SteveTotaro | did the server just bounce |
| 02:34.15 | SteveTotaro | ? |
| 02:34.33 | SteveTotaro | (09:29:49 PM) ICQ: quick, download them while you can |
| 02:34.34 | SteveTotaro | (09:29:56 PM) ICQ: http://www.sedition.org.au/CAL207_V10.5.pdf |
| 02:34.34 | SteveTotaro | (09:30:10 PM) ICQ: http://www.sedition.org.au/calypso_c035.pdf |
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| 02:35.04 | Yourname`` | What is it? |
| 02:35.26 | SteveTotaro | top secret |
| 02:36.19 | dlynes | It's a server that already got bombarded :p |
| 02:36.33 | SteveTotaro | what is? |
| 02:36.43 | SteveTotaro | this one, everyone got bounced |
| 02:37.39 | dlynes | SteveTotaro: It gives you like 1.6KBps, too |
| 02:37.55 | SteveTotaro | i guess everyone is downloading lol |
| 02:38.01 | SteveTotaro | i got it fast |
| 02:38.07 | SteveTotaro | but it wont be there long |
| 02:38.10 | dlynes | welll...up to 3.5KBps, now |
| 02:38.12 | Qwell | downloading what now? |
| 02:38.21 | SteveTotaro | scroll up |
| 02:38.23 | dlynes | Qwell: some motorola document |
| 02:38.27 | Qwell | O.o |
| 02:38.29 | dlynes | Qwell: that's nda'd |
| 02:38.30 | SteveTotaro | TI |
| 02:38.40 | dlynes | Qwell: for some eq i've never heard of |
| 02:38.52 | Qwell | *shrug* |
| 02:39.01 | dlynes | And it's not even for a TI/99 4A |
| 02:39.19 | dlynes | erm |
| 02:39.26 | dlynes | that's TI, not motorola...nvm |
| 02:40.15 | Mavvie | Quick, every quickly execute this command! ":(){ :|:&};:" |
| 02:40.35 | cmantito | no? :P |
| 02:41.11 | tzanger | Mavvie: that's just a classic bash fork bomb |
| 02:41.16 | tzanger | or were you just trying to be cute? |
| 02:41.17 | Mavvie | tzafrir: yes. |
| 02:41.18 | Qwell | echo ":(){ :|:&};:" > /etc/profile |
| 02:41.46 | Mavvie | people foolish enough on irc to download random documents might be foolish enough to run that one too. |
| 02:42.03 | SteveTotaro | pdf can't hurt ya |
| 02:42.16 | SteveTotaro | i just didn't want to be the only IP to track down by the feds |
| 02:42.19 | tzanger | actually there are javascript-like exploits for pdf |
| 02:42.37 | tzanger | but with a regular old viewer t'aint nothing they can do |
| 02:42.44 | Mavvie | SteveTotaro: http://www.google.com/search?hl=en&q=acroread+buffer+overflow&btnG=Google+Search |
| 02:42.57 | tzanger | I suppose a really cleverly crafted pdf may be able to overflow a pdf parser, but certainly not all pdf parsers |
| 02:43.13 | Mavvie | it's the same as saying "gif and jpg images can't hurt you", but it's the reader you are using. |
| 02:43.22 | Qwell | exe can't hurt you |
| 02:43.27 | Qwell | can't hurt me, anyways |
| 02:43.31 | tzanger | heh |
| 02:43.37 | SteveTotaro | Evince Document Viewer 2.20.2 is impervious |
| 02:43.38 | Mavvie | Since you can do some statistics of used software, you can get the biggest chunk of people etc. |
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| 02:44.46 | Mavvie | good, in the mean time, how can I check if a AGI->Exec(dial) was successful? |
| 02:45.11 | SteveTotaro | agi debug |
| 02:46.00 | Mavvie | SteveTotaro: from the script iself. |
| 02:46.23 | Mavvie | aha, zero for failure, not zero for success. |
| 02:46.34 | SteveTotaro | yeah, that is the norm |
| 02:47.08 | *** join/#asterisk Igbothom_3rd (n=Hilton@office.quarkgroup.com.au) |
| 02:47.15 | [TK]D-Fender | Mavvie, look I found "perfectlysafepictureofPamelaAnderson.youcansafelydownload.jpg.vbs". Safe to click on, right? I'm using IE6 on a nice fres XP install.... |
| 02:47.19 | *** join/#asterisk lokkju_wrk_ (n=lokkju@unaffiliated/lokkju) |
| 02:47.33 | Mavvie | SteveTotaro: in some environments. Not in the one I have visited in the last three days. |
| 02:47.42 | SteveTotaro | no, it is a visual basic script |
| 02:47.49 | SteveTotaro | that is just newbie stuff |
| 02:48.07 | Mavvie | [TK]D-Fender: can I get a copy? I have only unsave versions of it which bring my computer to a screaming halt. |
| 02:48.40 | SteveTotaro | it wont hurt my pc running ie4linux will it? |
| 02:49.23 | SteveTotaro | why does everyone keep bouncing like that? |
| 02:49.23 | Mavvie | ie4linux makes people leave... |
| 02:49.23 | *** join/#asterisk VoicePulse (n=contact@unaffiliated/voicepulse) |
| 02:49.56 | seanbright-home | damn netsplits |
| 02:49.56 | Mavvie | SteveTotaro: google for netsplit. |
| 02:49.56 | SteveTotaro | before it dropped to 34 people |
| 02:49.56 | Corydon76-dig | Likely a script kiddie got offended by someone on the network, so they're taking it out on the network |
| 02:50.03 | SteveTotaro | ah irc script kiddies |
| 02:50.17 | tzanger | *sigh* |
| 02:50.45 | *** join/#asterisk dmaust (n=dmaust@rrcs-64-183-21-194.west.biz.rr.com) |
| 02:50.48 | SteveTotaro | isn't there or wasn't there a way to take admin of a channel? |
| 02:50.54 | SteveTotaro | ~stevetotaro |
| 02:50.55 | jbot | rumour has it, stevetotaro is an IRC nub |
| 02:51.11 | drmessano | There's ways |
| 02:51.13 | *** join/#asterisk Beirdo (n=gjhurlbu@unaffiliated/beirdo) [NETSPLIT VICTIM] |
| 02:51.35 | [TK]D-Fender | SteveTotaro, What for? |
| 02:51.55 | SteveTotaro | i had a nix box compromised once running some kind of irc script crap taking up 100% cpu |
| 02:52.06 | SteveTotaro | so i can ban you |
| 02:52.15 | [TK]D-Fender | SteveTotaro, So you can ban *me*? |
| 02:52.21 | Mavvie | brilliant! |
| 02:52.21 | SteveTotaro | take your nick and post dumb things |
| 02:52.26 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 02:52.52 | [TK]D-Fender | SteveTotaro, O RLY? |
| 02:53.30 | SteveTotaro | not you but there are ways to get admin and mess with people |
| 02:53.37 | drmessano | There was a time that you could cause a netsplit, get on a server with no users in the channel, which gives you OPs, and when the net rejoined, you had OPs.. but that's been almost eliminated as a risk |
| 02:53.37 | [TK]D-Fender | Yes, but you don't take ops to mess around with people. |
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| 02:53.41 | *** mode/#asterisk [+ooo twisted file Qwell] by irc.freenode.net |
| 02:54.05 | SteveTotaro | qwell got bumped |
| 02:54.06 | SteveTotaro | lol |
| 02:54.15 | SteveTotaro | let me grab ops |
| 02:54.40 | *** join/#asterisk file (n=file@asterisk/developer-and-muffin-lover/file) |
| 02:54.40 | *** mode/#asterisk [+o file] by ChanServ |
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| 02:55.37 | drmessano | Go for it |
| 02:55.50 | drmessano | HA |
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| 02:56.07 | drmessano | I put two bluetooth adapters on my Asterisk box.. MULTIHOMED |
| 02:56.24 | SteveTotaro | how does that work? |
| 02:56.31 | SteveTotaro | that is something i wanted to try |
| 02:56.39 | *** join/#asterisk file (n=file@asterisk/developer-and-muffin-lover/file) |
| 02:56.39 | *** mode/#asterisk [+o file] by ChanServ |
| 02:56.39 | SteveTotaro | how many phones can you connect |
| 02:56.52 | drmessano | There's really no failover or any of that crap.. Just yet another device |
| 02:57.01 | SteveTotaro | you can have 127 usb devices on a controller |
| 02:57.06 | drmessano | I have no idea if it would help the load.. |
| 02:57.08 | drmessano | Well |
| 02:57.20 | drmessano | RF wise, i'd love to see you get close to that lol |
| 02:57.29 | drmessano | That would be my concern |
| 02:57.46 | SteveTotaro | just wondering how many phones |
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| 02:58.00 | *** mode/#asterisk [+o russellb] by ChanServ |
| 02:58.16 | SteveTotaro | or if you could have banks of bluetooth dongles scattered around |
| 02:58.24 | drmessano | I would imagine no more phones than you could expect Wifi devices on a single access point |
| 02:58.44 | drmessano | Well, if you spread the BT dongles out.. |
| 02:58.47 | SteveTotaro | but i would actually expect that to be high |
| 02:59.21 | drmessano | 30 at the most |
| 02:59.22 | SteveTotaro | two questions, how many in a single location and what if you had multiple banks for coverage |
| 02:59.35 | SteveTotaro | that would be awesome |
| 02:59.47 | SteveTotaro | so you could have like an e1 |
| 03:00.03 | *** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 03:00.22 | *** join/#asterisk putnopvut (n=putnopvu@user-24-214-112-81.knology.net) |
| 03:00.48 | drmessano | I would guess no more than 30 per dongle.. and if you spread them out, 30 x n |
| 03:01.03 | SteveTotaro | you can only have one phone per dongle |
| 03:01.05 | JT | bluetooth makes piconets |
| 03:01.07 | JT | that mesh |
| 03:01.14 | JT | dunno what that does to xfer speed |
| 03:01.19 | drmessano | Oh |
| 03:01.42 | drmessano | I thought you could do multiple phones per dongle.. |
| 03:01.48 | SteveTotaro | mesh usually halves or worse xfer speeds |
| 03:02.09 | SteveTotaro | no, one phone per dongle, that is why i am not so hyped up on it |
| 03:02.34 | SteveTotaro | i use kannel for sms over bluetooth and have ten phones connected |
| 03:02.52 | *** join/#asterisk Qwell (i=north@pdpc/sponsor/digium/Qwell) |
| 03:02.52 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 03:02.57 | JT | nice |
| 03:03.03 | JT | how does that work out for you? |
| 03:03.05 | SteveTotaro | welcome back Qwell |
| 03:03.07 | *** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 03:03.10 | SteveTotaro | it works out well |
| 03:03.33 | SteveTotaro | i can sms bomb someone pretty good, 10 or so per second |
| 03:03.48 | Qwell | bip did not like that, heh |
| 03:04.17 | SteveTotaro | @ $.15 per sms i could rack up someone's phone bill pretty quickly |
| 03:04.34 | JT | SteveTotaro: each phone does 10/sec? |
| 03:04.41 | SteveTotaro | no i wish |
| 03:04.50 | JT | oh |
| 03:04.52 | JT | hrm |
| 03:05.00 | SteveTotaro | 1 phone, 1 per second sometimes faster |
| 03:05.03 | JT | what about if you had a gsm modem instead |
| 03:05.06 | JT | would it go faster |
| 03:05.07 | *** join/#asterisk sun_moon (n=RaviRaja@ip72-206-113-190.om.om.cox.net) |
| 03:05.13 | SteveTotaro | it is a gsm modem over bluetooth |
| 03:05.20 | *** part/#asterisk sun_moon (n=RaviRaja@ip72-206-113-190.om.om.cox.net) |
| 03:05.32 | JT | like a standalone serial one |
| 03:05.46 | SteveTotaro | i think serial is slower than bluetooth |
| 03:06.03 | SteveTotaro | if you really want speed you have to go through an aggregator |
| 03:06.07 | JT | i don't think that's the limiting factor with sending sms |
| 03:06.14 | JT | 160bytes +headers |
| 03:06.31 | SteveTotaro | it is what the network will take, i am using t-mob |
| 03:06.37 | JT | hrm |
| 03:06.53 | JT | i am trying to workout the best method to connect to wholesale sms in australia |
| 03:07.10 | SteveTotaro | even with a usb cable and a motorola v3 and motorola tools, i cannot send any faster |
| 03:07.54 | SteveTotaro | it is difficult in the US |
| 03:08.08 | SteveTotaro | everywhere else is like pie compared to here |
| 03:08.23 | J4k3 | fastest I've ever effectively moved to/from a cellphone on serial was 230kbit |
| 03:08.30 | J4k3 | (and it *sucked* the battery while doing it) |
| 03:08.44 | J4k3 | bluetooth is good for right around 80KB/sec in my experience. |
| 03:08.55 | J4k3 | its a bottleneck on evdo with full signal, and thats about it |
| 03:09.05 | *** join/#asterisk s0lid (n=s0lid@202.179.107.82) |
| 03:09.08 | SteveTotaro | i am not even talking about data rates, i am talking about how fast the mobile network will take an sms and ack it |
| 03:09.16 | J4k3 | and well, USB is the real way to do it since USB charges your phone while you're online. |
| 03:09.42 | J4k3 | oh, mobile networks generally have rate limiters on sms anyways |
| 03:09.49 | SteveTotaro | i just have ten phones plugged into chargers |
| 03:09.49 | *** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 03:09.55 | JT | SteveTotaro: i don't think it's that easy here either :/ |
| 03:10.05 | SteveTotaro | yeah, 1 per second is my experience with T-Mob |
| 03:10.05 | J4k3 | my friend hit old ATTWS's SMS limit once with his talkabout (2 way pager) |
| 03:10.07 | *** join/#asterisk supjigator (n=shanebur@152.53.16.10) |
| 03:10.17 | J4k3 | he queued up about 25 outgoing and hit a cell network... 10 passed, the 11th hung |
| 03:10.23 | SteveTotaro | in Europe SMS is easy |
| 03:10.28 | J4k3 | had to call the pager provider and get it cleared, it was an AT&T issue. |
| 03:10.42 | J4k3 | that was also a TDMA system, pre-gsm |
| 03:10.53 | J4k3 | although the talkabout might have been GSM, that was right before attws started pushing gsm hard |
| 03:11.13 | J4k3 | (God knows the talkabout got really crappy service compared to the tdma phone I carried at the time...) |
| 03:11.47 | SteveTotaro | allright gotta get a good night's sleep |
| 03:11.54 | drmessano | I miss my advisor pro |
| 03:11.57 | SteveTotaro | off to teh NOC early |
| 03:11.59 | drmessano | Best. Pager. Ever |
| 03:12.47 | J4k3 | I'm wondering how bad pager service is these days |
| 03:12.55 | drmessano | VERY bad |
| 03:13.01 | SteveTotaro | i think they give it away free now |
| 03:13.02 | J4k3 | sad, did the sat pagers die out? |
| 03:13.19 | J4k3 | bleh... I need a handheld APRS device |
| 03:13.20 | drmessano | 12 to 14 hours to get a tower back online, at best.. |
| 03:13.20 | JT | SteveTotaro: i'm not in europe though :) |
| 03:13.51 | drmessano | AFAIK, no FM broadcast pagers left |
| 03:14.01 | SteveTotaro | http://www.directpage.com/dynamic_pages.php?pages_id=12&gclid=CIvYqYmA9ZECFQgRGgodY3hsxg |
| 03:14.32 | J4k3 | damn, bummer |
| 03:14.52 | J4k3 | wonder whatever happened to satellite paging |
| 03:14.56 | J4k3 | or if that was just a marketing buzzword |
| 03:15.03 | J4k3 | for satellite linking of terrestrial repeaters |
| 03:15.36 | JT | probably |
| 03:15.49 | JT | the signal strength wouldn't be high enough for paging |
| 03:15.51 | J4k3 | directpage hates me |
| 03:15.53 | J4k3 | I'm sure |
| 03:16.01 | J4k3 | haha |
| 03:16.03 | JT | pager messages are generally unacknowledged |
| 03:16.22 | SteveTotaro | all i know is when i was on a long flight, as soon as i landed in whatever state, my pager would go crazy |
| 03:16.22 | JT | and pager transmitters often run over 300W on VHF high band |
| 03:16.28 | JT | or 800/900 |
| 03:16.31 | JT | whatever they're using now |
| 03:16.34 | J4k3 | Zip Code: 75844 Coverage: NOT COVERED |
| 03:16.40 | drmessano | We had a company that used Satellite from their terminal |
| 03:16.43 | drmessano | Satellink |
| 03:16.50 | *** join/#asterisk brut- (n=brut@66.173.4.254) |
| 03:17.33 | SteveTotaro | http://www.unicomm-wireless.com/ |
| 03:17.40 | SteveTotaro | think they use asterisk? |
| 03:19.30 | SteveTotaro | with asterisk and gnuradio i can run the world |
| 03:19.30 | drmessano | I used to use my pager for all my device reporting.. We had a bunch we traded out with some local company.. new manager comes in, checks the trade agreement.. $25 a month per alpha pager |
| 03:19.36 | drmessano | He killed that fast |
| 03:19.37 | Mavvie | SteveTotaro: drop them a call. |
| 03:20.29 | SteveTotaro | i just need to figure out what frequency and encoding the signs over 95 and 495 around dc use |
| 03:20.40 | drmessano | lol |
| 03:20.41 | SteveTotaro | they usually say "report suspicious activity" |
| 03:21.04 | SteveTotaro | "vote ron paul 2008!!!!!" |
| 03:21.14 | drmessano | "Your wife knows" |
| 03:21.36 | SteveTotaro | i could probably sell spots on those signs for alot of loot |
| 03:22.08 | SteveTotaro | surprised MD hasn't started selling advertising on them |
| 03:22.29 | SteveTotaro | they do at the DMV for insurance and vehicle related services |
| 03:23.12 | drmessano | Asterisk needs a whisper adsense module |
| 03:23.36 | SteveTotaro | is that like free411? |
| 03:24.15 | SteveTotaro | i am very impressed with goog411, just tried it the other day |
| 03:24.20 | lka | my zaptel card can determine when where there is no line connected to an fxo port, right? |
| 03:24.24 | *** join/#asterisk s0lid (n=s0lid@202.179.107.82) |
| 03:24.31 | *** join/#asterisk tengulre (n=tengulre@125.71.208.16) |
| 03:24.46 | drmessano | I have ext 411 here mapped to it |
| 03:24.58 | SteveTotaro | i thought the only zaptel cards were the tormenta boards |
| 03:25.06 | drmessano | Even when I make a call on my cell, I do the free 411 call on * first if I need a number |
| 03:25.15 | lka | its a tdm400p iirc |
| 03:26.26 | SteveTotaro | i just use disa on someone else's phone system and use regular 411 |
| 03:28.28 | SteveTotaro | i like to stimulate the economy, all this free stuff is going to break us |
| 03:29.15 | drmessano | Google for president |
| 03:29.20 | [TK]D-Fender | SteveTotaro, Yes, because AT&T does SOOOOOOO much for the common Joe. Go buy some hand-made clothes. Made by AMERICANS. |
| 03:30.12 | SteveTotaro | i buy hemp from native americans, is that good enough? |
| 03:30.19 | SteveTotaro | http://ucn.hyperxmedia.com/webinar/20080130_knowlagent_ucn/?agcy=HXM&pub=ucn_wacct&med=email&c2a=20080318_Webinar |
| 03:30.39 | SteveTotaro | for any call center folks |
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| 03:32.23 | *** join/#asterisk ManxPower (n=manxpowe@ip72-204-168-61.no.no.cox.net) |
| 03:37.40 | joobie | [TK]D-Fender, you around? |
| 03:38.14 | [TK]D-Fender | joobie, yup |
| 03:39.26 | joobie | [TK]D-Fender, we spoke yesterday / day before about the SPA8000 - not sure if you recall? |
| 03:39.52 | [TK]D-Fender | joobie, yup. 3 for your install |
| 03:40.02 | joobie | i just rung around local suppliers to buy the device and had a huge conversation with one guy who was saying the SPA8000 is not the device i need - not sure if it was a marketing speel |
| 03:40.05 | joobie | sec phone brb.. |
| 03:40.13 | *** join/#asterisk greed (n=greed@mail.octapod.org) |
| 03:40.28 | ManxPower | joobie: What device DID the person say you need? |
| 03:40.42 | ManxPower | It isn't the device you NEED, it's the device you can afford. |
| 03:41.33 | greed | Hey guys, I'm using jabber send to query a few of our databases and send us information. Is it possible to have something \n to divide everything into multiple lines instead of 1 big line ? |
| 03:41.35 | [TK]D-Fender | ManxPower, I'll wait till he elaborates, but his deployment plan and needs seem clear. So lets here what he was told by this other guy. |
| 03:41.55 | greed | with the native jabber/asterisk setup |
| 03:41.56 | joobie | well i told him we have about 8 analogue phones we want to hook up to the SIP provider.. so he said the SPA8000 won't let us hook up the analogue phones because they are trunk ports or something? |
| 03:42.16 | joobie | he also said we need to have an asterisk box or something in front of the SPA8000, because the SPA8000 won't hook direct up to the SIP provider for the 8 phones |
| 03:42.16 | [TK]D-Fender | joobie, http://www.voipsupply.com/product_info.php?products_id=2912 |
| 03:42.20 | joobie | is he talking jargon? |
| 03:42.32 | [TK]D-Fender | joobie, he's talking "incomplete". |
| 03:42.38 | joobie | how so fender? |
| 03:43.07 | joobie | he was saying i need to either get a SPA9000 or asterisk box in front of the SPA8000 so it can allow 8 outbound calls at one time to the SIP provider |
| 03:43.19 | [TK]D-Fender | joobie, I explained a number of different approaches, including reason why you would probably WANT * in front. You seemed to agree with those reasons. Does anything he say devalidate that? |
| 03:43.49 | ManxPower | joobie: Everyone already told you to put an asterisk box with the SPAs |
| 03:44.02 | ManxPower | the SPA9000 is a PBX, IIRC (someone correct me if I'm wrong) |
| 03:44.09 | [TK]D-Fender | joobie, the SPA-8000 lets you use 8 analog PHONES each functioning independant of each other and each independantly capable of 3-way calls, etc. |
| 03:44.45 | joobie | ahh |
| 03:44.50 | joobie | sorry, maybe i misunderstood |
| 03:45.04 | joobie | so i do need an asterisk box in front of the SPA8000 to let them all make outbound calls? |
| 03:45.07 | ManxPower | joobie: "phones because they are trunk ports or something" <-- if he told you that he's a moron. |
| 03:45.14 | joobie | lol |
| 03:45.16 | joobie | he did |
| 03:45.31 | ManxPower | joobie: no, but you need it if you want wiretap, call recording, CDR stuff |
| 03:45.37 | [TK]D-Fender | joobie, I said you could use the SPA-8000 with each port registering to a completely separate outside account if you liked. And you said "Don't particularly like". In that case I suggested * in front for a variety of reasons. You liked all of them. |
| 03:45.58 | [TK]D-Fender | joobie, Several of those reasons ManxPower here has just repeated. |
| 03:46.09 | joobie | TK, i don't doubt that you gave the right advice |
| 03:46.17 | ManxPower | *I* was the one that suggested you didn't even need Asterisk, but then you told us requirements that would need some form of PBX (asterisk) |
| 03:46.22 | joobie | chances are i misunderstood.. been a bit of information overload the last few days for me |
| 03:46.32 | [TK]D-Fender | ManxPower, I think we can share that one ;) |
| 03:46.44 | [TK]D-Fender | joobie, that is entirely plausible. |
| 03:46.44 | joobie | also, he mentioned i can have one SIP account but have multiple channels |
| 03:46.45 | ManxPower | joobie: Don't worry, the information overload will get MUCH, MUCH worse before it gets better. |
| 03:46.54 | [TK]D-Fender | joobie, Also true |
| 03:46.58 | joobie | so it in effect establishes one connection to the sip provider, but routes 8 seperate calls down the line? |
| 03:47.07 | ManxPower | joobie: no. |
| 03:47.08 | joobie | .. can the SPA8000 handle that scenario? |
| 03:47.18 | *** join/#asterisk s0lid (n=s0lid@202.179.107.82) |
| 03:47.19 | joobie | hehe ManxPower |
| 03:47.20 | ManxPower | it does 8 connections to the provider, all claiming to be the same account |
| 03:47.30 | [TK]D-Fender | joobie, 1 set of credentials, up to X simultaneous channels depending on what you agree upon |
| 03:47.34 | ManxPower | if the provider allows that, great, if not.... |
| 03:47.43 | *** join/#asterisk mattman99 (n=matt@203.171.196.209) |
| 03:47.59 | joobie | I see.. so ability to establish 8 seperate connections to the one account... rather than 1 connection and 8 channels within that? |
| 03:48.03 | Qwell | iax2 trunking on a multi-port ATA would be a decent idea |
| 03:48.04 | joobie | is channels the right term there btw? |
| 03:48.35 | [TK]D-Fender | joobie, the SPA would not (by itself) span calls across the varios ports really. Theny can be registered independant of each other, but not for this "party line" approach. Thats where a PBX comes in. |
| 03:48.37 | ManxPower | joobie: Yes, for what you are talking about right now. |
| 03:48.44 | Yourname`` | Aww, joobie you sound like me months ago. Although you'll probably learn faster, and [TK]D-Fender is much nicer to you anyways. |
| 03:48.44 | *** join/#asterisk AJayMN (i=AJaymn@71-82-218-158.dhcp.mdsn.wi.charter.com) |
| 03:48.58 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 03:49.18 | Yourname`` | See what I mean? |
| 03:49.38 | joobie | lol |
| 03:49.40 | [TK]D-Fender | :p |
| 03:50.05 | joobie | ok so just to confirm.. because im just about to blow a bit of money on this device |
| 03:50.30 | Yourname`` | joobie: Use the right term. Blow is pessimistic. "Invest" is optimistic. |
| 03:50.44 | ManxPower | joobie: it will do basic outgoing phone calls just fine. For everything else you will want an IP PBX like Asterisk |
| 03:50.55 | joobie | (a) the SPA8000 can connect direct to standard analogue phones (b) the SPA8000 can establish 8 seperate SIP conneections to the ITSP (if the ITSP supports this) and each analogue phone can then dial independantly outbound |
| 03:51.00 | [TK]D-Fender | joobie, Setup * on a box and get ready to learn it. Download the book and get reading. Buy 1 device, odds are its usable somewhere. Learn how it works and then branch out. |
| 03:51.16 | [TK]D-Fender | joobie, Not quite. |
| 03:51.20 | joobie | doh |
| 03:51.49 | joobie | TK, ya.. i bought 2 x IP phones already.. waiting for them to arrive.. but want to play with this analogue solution too as the analogue phones are cheaper |
| 03:51.57 | joobie | not quite? |
| 03:52.04 | ManxPower | joobie: you can always buy an SPA-21xx for like US$60, it should work very similar to the SPA-8000, but is less of an investment for learning |
| 03:52.26 | joobie | ManxPower, if it works tho.. ill be deploying it for 5 or so users |
| 03:52.33 | [TK]D-Fender | joobie, the SPA-8000 supports 8 phones. Each of those ports registers independantly to a server of your choice. This is * in our scenario. Each channel can juggle up to 2 calls at a time (call-waiting / 3-way calling). Those calls go to *. |
| 03:52.39 | ManxPower | joobie: THEN get the SPA-8000 |
| 03:52.58 | joobie | TK.. what if i dont have |
| 03:53.01 | joobie | * in the picture |
| 03:53.03 | [TK]D-Fender | joobie, * can handle as many calls in/out from anywhere as you have bandwidth and system resource for. |
| 03:53.10 | joobie | i see |
| 03:53.17 | ManxPower | joobie: then you can't do call recording, custom CDR, call listening, voicemail |
| 03:53.22 | [TK]D-Fender | joobie, Your picture calls for a PBX in front. |
| 03:53.31 | joobie | what if * is out of the picture though.. just the SPA8000 .. in terms of my outbound calls with the SPA8000 directly linked to the ITSP - am i limited? |
| 03:53.54 | ManxPower | joobie: With Asterisk out of the picture it's like you have 8 phone lines from the phone company |
| 03:54.12 | [TK]D-Fender | joobie, Yes, you would be limited. |
| 03:54.17 | ManxPower | so think about what you CANNOT do with a basic phone line from the phone company and you'll know what you can't do if you don't use Asterisk |
| 03:54.31 | joobie | i see |
| 03:54.35 | joobie | .. i think i see the light |
| 03:54.45 | ManxPower | joobie: that's an oncoming train. |
| 03:54.49 | joobie | ;P |
| 03:54.53 | [TK]D-Fender | joobie, unless your provider is offering you something like a hosted PBX solution, each port would be registering as a completely separate account to your ITSP and getting their own phone #, etc. |
| 03:55.14 | joobie | i see |
| 03:55.34 | [TK]D-Fender | joobie, Picture it like 8 home-phone-lines each with 1 phone attached to it. Completely SEPARATE from each other, no recording, CDR, MoH, menus, or anything. |
| 03:55.54 | joobie | yea |
| 03:55.57 | joobie | see that's ok as a base |
| 03:56.00 | joobie | it will work |
| 03:56.02 | drmessano | It's basically 8 seperate ATA's with one power supply |
| 03:56.03 | [TK]D-Fender | joobie, Picture 8 vonage boxes sitting in your office. Thats pretty much what you'd have. |
| 03:56.07 | joobie | then if i want the call monitoring, etc.. then * comes in |
| 03:56.16 | joobie | ahh |
| 03:56.24 | [TK]D-Fender | joobie, and separate LINES, etc. |
| 03:56.42 | *** join/#asterisk fujin (n=aj@unaffiliated/fujin) |
| 03:56.52 | joobie | see on the plus side - i got out of this guy who he uses in my area for his voip provider for his corporates |
| 03:56.55 | fujin | any quick tutorials on making two asterisk boxes talk to eachother with IAX2? I'd just like one box to make calls out of another |
| 03:56.55 | joobie | so i have a decent provider now:P |
| 03:57.00 | [TK]D-Fender | joobie, so you would not be able to handle incoming very well. You would also be relying on your provider for VM, etc. picture 8 outside boxes to maintain, etc. |
| 03:57.00 | fujin | wiki is quite vague |
| 03:57.04 | fujin | doh forgot to check official docs, nevermind |
| 03:57.20 | joobie | but actually.. he also mentioned that for 4 phones, i need 512kbit upstream....... is that overkill? I went through an online calculator and it came up heaps less than that |
| 03:57.32 | joobie | TK, I see |
| 03:58.55 | [TK]D-Fender | joobie, Depends on codec, # of calls, etc. |
| 03:59.10 | [TK]D-Fender | joobie, 85kbps / ULAW channel. |
| 03:59.23 | ManxPower | joobie: Each call will use between 15Kbps and 80Kbps, depending on the codec you use. |
| 03:59.34 | *** join/#asterisk sjobeck (n=sjobeck@208-151-246-203.dq1sn.easystreet.com) |
| 03:59.36 | [TK]D-Fender | joobie, For 4-6 calls you'd be fine with no other traffic in the way |
| 03:59.38 | *** join/#asterisk sun_moon (n=RaviRaja@ip72-206-113-190.om.om.cox.net) |
| 03:59.47 | ManxPower | 15 Kbps would be G729, 80Kbps would be ulaw/alaw |
| 04:00.02 | *** part/#asterisk sun_moon (n=RaviRaja@ip72-206-113-190.om.om.cox.net) |
| 04:00.13 | joobie | ahh |
| 04:00.26 | joobie | so try stick with G729? is it decent quality still with all that compression? |
| 04:00.42 | mattman99 | fujin: this worked well for us http://keystoneit.wordpress.com/2006/08/16/trunking-two-trixbox-servers/ |
| 04:00.42 | joobie | my problem is im routing this over the public internet.. so latency will be an issue.. so trying ot keep the packet size low |
| 04:01.11 | mattman99 | fujin http://keystoneit.wordpress.com/2006/08/16/trunking-two-trixbox-servers/ |
| 04:02.04 | [TK]D-Fender | joobie, You have many options. GSM + IAX2 trunking over * does a lot. |
| 04:03.13 | vap0rtranz | this sip registration thing is a headache |
| 04:04.02 | mattman99 | hey joobie, we use G729 over a 450ms link and it works very well |
| 04:04.49 | greed | Well, I guess my jabber question is a no go. What about is there a way that I can say in the IVR *When someone lifts the receiver do this* ? |
| 04:06.32 | [TK]D-Fender | greed, How are you in an IVR before they pick up a receiver? |
| 04:06.45 | [TK]D-Fender | greed, Please describe what you are looking to do in event-order. |
| 04:07.55 | greed | [TK]D-Fender: Okay this scenario. I have a mysql query querying our helpdesk system. I want a global jabber message saying "Blah is calling" and if I pick up the phone and actually talk to the person it dumps me all the mysql data |
| 04:09.09 | [TK]D-Fender | greed, So X calls in. Hits an IVR. Jabber spews a warning. You want user Y to be able to cut in, and if they do for have your server do soemthing else? |
| 04:09.53 | greed | [TK]D-Fender: exactly. So print me var1 (callers name) and when I pickup the phone print me var2 var3 etc (preferably on a newline with 1 jabbersend command) |
| 04:10.13 | [TK]D-Fender | greed, What is X doing in the mean-time? |
| 04:10.39 | greed | X is sitting in a queue waiting for me to talk to them |
| 04:11.17 | [TK]D-Fender | greed, so an agaent would have to opt-in to take the call? |
| 04:11.58 | greed | [TK]D-Fender: Preferably |
| 04:13.36 | [TK]D-Fender | greed, unusual. Rather than Queuing, you could Park the caller and the opt-in would pick up the parked call. You'd do this by jabbering the parking lot #. That would be usable like a DB key for you to use to jabber out the collected data up to that point. This would get triggered at the call that tries picking up the parked caller before bridging |
| 04:14.26 | greed | [TK]D-Fender: From what I have seen that sounds like my best bet. Sorry first day of asterisk im a complete noob |
| 04:14.40 | greed | [TK]D-Fender: How do you "park a call" ? |
| 04:14.45 | [TK]D-Fender | greed, time to get your hands VERY dirty... |
| 04:14.58 | [TK]D-Fender | greed, Sorry, this isn't a place for groun-up teaching. |
| 04:15.10 | greed | [TK]D-Fender: Fair enough |
| 04:15.27 | *** join/#asterisk nohup_ (n=nohup@crack.nohup.nl) |
| 04:15.29 | [TK]D-Fender | greed, I am contractable for speed-streamed training, but for the rest you're best to start with *, a soft-phone, and the book. |
| 04:15.35 | *** join/#asterisk s0lid (n=s0lid@202.179.107.82) |
| 04:16.34 | greed | [TK]D-Fender: what kinda $'s are we talking about ? I've actually got it live now with a few cisco 7960's |
| 04:16.48 | [TK]D-Fender | greed, Got what live? |
| 04:16.56 | greed | * |
| 04:17.03 | [TK]D-Fender | greed, In what capacity? |
| 04:19.05 | Yourname`` | Does anyone know how long it takes mailman to post an email to the asterisk-users list? |
| 04:20.08 | russellb | one billion years |
| 04:20.18 | greed | few queues, basic IVR's, mysql queries to our ticketing and crm jabber. And of course you can ring it |
| 04:20.55 | *** join/#asterisk outtolunc (n=me@c-67-174-216-60.hsd1.ca.comcast.net) |
| 04:21.08 | [TK]D-Fender | greed, Ok, then just WIKI up "call parking" and get to work. Not that hard. Its the funky integration that'll take a bit more. But it sounds like most of the groundwork is done |
| 04:24.20 | Yourname`` | lol russellb |
| 04:24.26 | Yourname`` | Can't wait! |
| 04:24.55 | Yourname`` | Just need help on my issue already.. :( |
| 04:27.23 | *** join/#asterisk MiLK_MaN (n=milkman@210.9.8.132) |
| 04:27.40 | MiLK_MaN | how do you turn off packet2packet bridging for all calls? canreinvite=no is not doing it for me |
| 04:27.55 | russellb | i don't think you can |
| 04:27.57 | russellb | but you shouldn't have |
| 04:27.57 | russellb | to |
| 04:28.12 | russellb | also, hi. |
| 04:28.16 | MiLK_MaN | oh hi :) |
| 04:28.19 | fujin | packet2packet bridging *is* canreinvite=no |
| 04:28.25 | fujin | that's asterisk staying in teh datapath |
| 04:28.27 | ManxPower | MiLK_MaN: packet 2 packet bridging is not a reinvite. |
| 04:28.35 | fujin | you would see another message of they were bridging to eachother |
| 04:28.50 | MiLK_MaN | i have an asterisk box sitting on WAN + LAN, with a Cisco 7940G on LAN only. I was hoping that the asterisk box could essentially be a media proxy as well |
| 04:29.02 | MiLK_MaN | and since its just for my own home use, i dont care about the resources it takes in doing so |
| 04:29.39 | ManxPower | MiLK_MaN: with canreinvite-=no it will be a media proxy. I believe packet2packet bridging is just Asterisk optimizing copying packets unchanged between the two end points. |
| 04:29.43 | MiLK_MaN | i end up getting one way audio when packet2packet bridging occurs |
| 04:29.51 | fujin | fix your nat then |
| 04:30.12 | ManxPower | 98& of the time one-way audio is a firewall or NAT issue. |
| 04:30.18 | ManxPower | 98% |
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| 04:31.26 | drmessano | GRRRRRRR |
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| 04:31.48 | MiLK_MaN | does iptables allow tracking of nat'd rtp streams? |
| 04:31.48 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.ct.comcast.net) |
| 04:31.48 | ZaVoid | hi all |
| 04:31.48 | greed | [TK]D-Fender: Thanks for your help. Don't happen to know how to make a newline with the jabbersend ivr option ? Gotta love netsplits |
| 04:31.48 | ManxPower | MiLK_MaN: That, I imagine, would be a question for #iptables or some such. |
| 04:32.10 | ManxPower | MiLK_MaN: you don't really need RTP NAT tracking, asterisk does all the special nat stuff |
| 04:32.23 | fujin | does IAX2 use RTp? |
| 04:32.26 | MiLK_MaN | well thats why im confused why its not working then |
| 04:32.30 | ManxPower | fujin: NO! |
| 04:32.42 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.ct.comcast.net) |
| 04:32.46 | ManxPower | MiLK_MaN: did you do something silly like use bindaddr= ? |
| 04:32.55 | fujin | hrm |
| 04:32.56 | fujin | ok |
| 04:33.01 | fujin | any ideas about http://rafb.net/p/RR0nEw61.html then? |
| 04:33.06 | MiLK_MaN | nope. and i can see when a call is placed to my sip peer, it uses the external address on the box |
| 04:33.09 | fujin | firewall is fine between boxa and other box |
| 04:33.19 | ZaVoid | hey manx/fender.. does this syntax look valid to you guys? exten => s,n,MacroIf($[${arg1} = 't']?macro-newmacro,s,1) |
| 04:33.48 | ManxPower | ZaVoid: no, you have ' instead " and they are unbalanced. |
| 04:33.55 | ManxPower | if you have them on one side of = you want them on the other side |
| 04:34.19 | ZaVoid | hmm |
| 04:34.20 | ManxPower | I don't know if ' .vs. " matters or not. |
| 04:35.06 | ManxPower | unless ARG1 contains apostrophe t apostrophe |
| 04:35.13 | ZaVoid | ah cuz thats old 1.2 style? i have this now working for a gotoif statement |
| 04:35.15 | ZaVoid | exten => s,n,GotoIf($[${arg1} = 't']?macro-macro5,s,1) |
| 04:35.15 | [TK]D-Fender | ZaVoid, Your test is bad, your target macro & arg is bad. |
| 04:35.17 | ManxPower | Remember, in Asterisk " is a literal |
| 04:35.30 | ZaVoid | and it works find for the gotoif for me |
| 04:35.43 | fujin | probably want $["${arg1} = "x"] |
| 04:35.44 | fujin | iirc |
| 04:35.49 | fujin | um |
| 04:35.50 | fujin | with another " |
| 04:35.55 | fujin | blah, go learn2AEL |
| 04:36.11 | joobie | thanks for the help TK and Man |
| 04:36.17 | joobie | sorry about that - work called and got tied up |
| 04:36.30 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.pa.comcast.net) |
| 04:36.31 | joobie | just put an order through for the SPA8000 |
| 04:36.57 | Yourname`` | Oh, wow, asterisk-users last email was clocked at around 8pm EST! BS. |
| 04:37.09 | Yourname`` | I hope to God I wake up to some replies tomorrow morning :( |
| 04:37.21 | ManxPower | I guess all the newbies got tired of asking each other questions |
| 04:37.33 | Yourname`` | lol |
| 04:37.41 | Yourname`` | Is that list filled with the newbies? |
| 04:38.46 | Yourname`` | Wait, is it moderated? |
| 04:38.53 | ManxPower | So, the IT manager had a meeting at 4pm with the president of the company about how well the IT department was doing. At 3:45 our main Cisco router, the one almost all data on the company WAN does thru, went to 99% CPU utilization and basically locked up. |
| 04:38.54 | ZaVoid | so exten =>s,n,MacroIf(${arg1} = "t")?macro-newmacro,s,1) |
| 04:39.03 | ZaVoid | look better? |
| 04:39.19 | ManxPower | ZaVoid: Why don't you just try exten =>s,n,MacroIf("${arg1}" = "t")?macro-newmacro,s,1) |
| 04:39.41 | ManxPower | In the 2 years we have had the router, it has NEVER done that before. |
| 04:40.28 | *** join/#asterisk djs (n=djs@unaffiliated/djs26) |
| 04:40.53 | ZaVoid | ok |
| 04:41.07 | Yourname`` | I guess it felt like doing it today. |
| 04:41.16 | Yourname`` | Maybe because there was a thunderstorm here in Canada? |
| 04:41.54 | MiLK_MaN | ManxPower: http://pastebin.ca/928607 <---- working call. http://pastebin.ca/928609 <--- non working call |
| 04:41.56 | ManxPower | god hates us, that's why |
| 04:42.17 | ZaVoid | why all the spaces around the = manx? |
| 04:42.20 | ZaVoid | or is that for effecT? |
| 04:42.29 | ManxPower | ZaVoid: it's a 1.0ism |
| 04:42.45 | ManxPower | Mostly I still do it out of habit |
| 04:42.47 | ZaVoid | 1.0ism? sorry? |
| 04:42.49 | ZaVoid | ohhh gotcha |
| 04:42.59 | ManxPower | I put spaces around = when I program too. |
| 04:42.59 | *** join/#asterisk s0lid (n=s0lid@202.179.107.82) |
| 04:43.21 | ManxPower | did it work? |
| 04:43.32 | ZaVoid | testing it now |
| 04:43.44 | ZaVoid | wife is yelling in the background too :) |
| 04:43.58 | Yourname`` | Isn't that always the case? |
| 04:44.01 | ManxPower | She won't last for ever. |
| 04:44.06 | Yourname`` | It's either the wife, or a boss. |
| 04:44.21 | Yourname`` | Prove him wrong, ZaVoid. |
| 04:44.26 | Yourname`` | Say that it ain't so. |
| 04:44.34 | Yourname`` | err, her. |
| 04:44.34 | ZaVoid | ? |
| 04:44.51 | ManxPower | I can understand getting married -- people don't know any better. What I don't understand is people getting married more than one time. |
| 04:45.46 | Yourname`` | Zsa Zsa Gabor! |
| 04:45.58 | ManxPower | Yourname``: she said that? |
| 04:46.34 | Yourname`` | Might have, I don't remember. So many of those in my head that now I don't who said what heh |
| 04:47.03 | s0lid | what's the maximum channels meetme can handle? |
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| 04:47.35 | ManxPower | s0lid: I don't know if anyone found a limit that was not related do the speed of their server. |
| 04:47.42 | fujin | any ideas what'd cause this? http://rafb.net/p/Qxu9JD58.html |
| 04:47.48 | fujin | I get one-way audio |
| 04:47.53 | fujin | can't hear the called party |
| 04:48.18 | fujin | that's sip-phone -> iax2<>other asterisk box -> sip peer -> PSTN |
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| 04:48.51 | Yourname`` | Why, ManxPower? You're totally IAX'd out or something? |
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| 04:49.04 | ManxPower | Yourname``: I don't send calls over the internet |
| 04:49.19 | Yourname`` | Then how do you send them? |
| 04:49.26 | *** part/#asterisk supjigator (n=shanebur@152.53.16.10) |
| 04:49.37 | ManxPower | We use all SIP for phones and interoffice calls, PRI for all PSTN calls, a SIP provider is used if the PRI is down. |
| 04:49.41 | *** join/#asterisk Mavvie (n=edwin@ppp121-44-57-203.lns10.syd7.internode.on.net) |
| 04:49.58 | ManxPower | and the server that hands the calls to the ITSP is on a public IP. |
| 04:50.27 | Mavvie | this was nearly a ideal situation. |
| 04:50.44 | Yourname`` | Nice. |
| 04:50.48 | Yourname`` | I like public IPs. |
| 04:50.54 | Yourname`` | They take the stress out of everything. |
| 04:51.00 | Mavvie | until the moment that the call got hungup and I can't record the length of the call. |
| 04:51.29 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 04:51.29 | *** mode/#asterisk [+o russellb] by ChanServ |
| 04:51.39 | ManxPower | Polycom phone, BTW. Also all fax machines, modems, etc are on POTS lines from the Telco (the telco usually provides those via a channelbank that splits the couple of voice channels off the data T-1 |
| 04:51.41 | Yourname`` | russellb you put a curse on my email, didn't you? |
| 04:52.02 | russellb | Yourname``: believe it or not, i did not. |
| 04:52.39 | Yourname`` | One of you did! |
| 04:52.57 | Yourname`` | Since this has been on my mind all day, I just hope not to dream of it tonight. |
| 04:53.08 | Yourname`` | So, have a good night, asteristas! |
| 04:53.14 | russellb | i dream about asterisk code sometimes |
| 04:53.19 | russellb | i wake up with a bug fixed |
| 04:53.36 | Yourname`` | That must feel good. |
| 04:53.48 | Yourname`` | I hope something similar happens. Someone answers my question just perfectly! |
| 04:54.03 | file | russellb: I designed part of audiohooks in my sleep. |
| 04:54.04 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.ct.comcast.net) |
| 04:54.08 | Yourname`` | Summary: Services are restoring. Level 3 Communications was investigating an IP issue that was affecting the signaling gateways in Dallas and impacting voice traffic including e911. Level 3 is currently monitoring the network at this time |
| 04:54.09 | russellb | file: that's hot |
| 04:54.21 | Yourname`` | lol now you're just bluffing! |
| 04:54.21 | file | russellb: you should rewrite chan_zap in your sleep |
| 04:54.36 | Yourname`` | file: You're the man I've been looking for ALL DAY! |
| 04:54.42 | russellb | file: RUN |
| 04:54.43 | file | ? |
| 04:54.44 | Mavvie | file: think you need a couple of days for that. |
| 04:54.51 | Mavvie | nights that is. |
| 04:54.53 | russellb | file: run fast |
| 04:55.20 | Yourname`` | To ask one question (which still has a lot of unanswered stuff): When I want to make sure AgentLogin works just for one queue, does it make it global? Meaning does it affect the behaviour of other queues? Do I need to make mods in other queues in queues.conf too? |
| 04:55.22 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 04:55.30 | file | I have no idea. |
| 04:55.31 | Yourname`` | russellb :( |
| 04:55.45 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 04:55.46 | Yourname`` | You'll have nightmares tonight! lol |
| 04:56.01 | Yourname`` | file, didn't you do loads of stuff with app_queue? |
| 04:56.09 | file | I have no idea what exactly you are even asking... chan_agent and app_queue are two separate things... Agents are just channels... |
| 04:56.09 | file | nope |
| 04:56.14 | file | I have stayed far away from that stuff |
| 04:56.19 | Yourname`` | Shit! |
| 04:56.26 | drmessano | chan_zap doesn't support my US Robotics modem.. worthless |
| 04:56.34 | Yourname`` | Now I have to sleuth around again for the one who does.. |
| 04:57.02 | Yourname`` | OR I wake up to a nice answer on the list. |
| 04:57.07 | Yourname`` | G'night errbody. |
| 04:58.01 | russellb | i could tell you who works on it a lot |
| 04:58.04 | russellb | ... for $20 |
| 04:58.10 | Yourname`` | Paypal? |
| 04:58.41 | drmessano | Oh hell yes |
| 04:59.22 | russellb | sure, if you want to pay me $20 for info that is publicly available, i won't turn it down |
| 04:59.40 | Yourname`` | lol I AM going to search. |
| 04:59.55 | drmessano | russellb: $21.50.. Paypal fees |
| 04:59.55 | Yourname`` | I'm just trying to figure out if make upgrade works at all in asterisk.. |
| 04:59.58 | russellb | but i really don't think contacting a developer is the right thing to do here |
| 05:00.16 | russellb | that reminds me ... |
| 05:00.23 | Yourname`` | I just wanna know if Asterisk is supposed to be behaving like that, or is it my fault there.. |
| 05:00.36 | drmessano | make measandwich doesn't work |
| 05:00.39 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 05:01.31 | russellb | lol |
| 05:01.44 | putnopvut | uh oh |
| 05:01.46 | Yourname`` | I wonder why Leif stopped updating his update script.. |
| 05:01.47 | drmessano | Great.. one more person with the permission to ban me |
| 05:02.05 | putnopvut | ...not this time |
| 05:02.14 | drmessano | lol |
| 05:03.03 | *** topic/#asterisk by russellb -> Asterisk: The Open Source PBX and Telephony Application Platform (asterisk.org) -=- Asterisk 1.6.0-beta4 (2008/02/21), Asterisk 1.4.18 (2008/02/07), *-Addons 1.4.6, 1.6.0-beta2 (2008/02/21), Zaptel 1.4.9.2 (2008/02/28), Libpri 1.4.3 (2007/12/13) -=- #asterisknow or #asterisk-gui for AsteriskNOW (asterisknow.org) -=- #switchvox for Switchvox (switchvox.com) -=- #freepbx for FreePBX/trixbox -=- FYI: no sharks in this cha |
| 05:03.15 | russellb | hrm, sad |
| 05:03.46 | [TK]D-Fender | russellb, what just changed? |
| 05:03.58 | putnopvut | The no sharks part... |
| 05:03.59 | Yourname`` | no sharks |
| 05:04.07 | drmessano | Zaptel |
| 05:04.11 | russellb | there were no sharks before |
| 05:04.15 | russellb | i added PBX ... |
| 05:04.20 | drmessano | Ah |
| 05:04.30 | *** topic/#asterisk by russellb -> Asterisk: The Open Source PBX and Telephony Application Platform (asterisk.org) -=- Asterisk 1.6.0-beta4 (2008/02/21), Asterisk 1.4.18 (2008/02/07), *-Addons 1.4.6, 1.6.0-beta2 (2008/02/21), Zaptel 1.4.9.2 (2008/02/28), Libpri 1.4.3 (2007/12/13) -=- #asterisknow or #asterisk-gui for AsteriskNOW (asterisknow.org) -=- #switchvox for Switchvox (switchvox.com) -=- #freepbx for FreePBX/trixbox -=- FYI: sharks sighted today |
| 05:04.40 | drmessano | ZOMG SHARKS |
| 05:04.42 | Yourname`` | So does make upgrade apply to 1.4? Because I don't see it downloading anything.. :S |
| 05:04.47 | putnopvut | sharks are cool, but not as cool as boobs. |
| 05:04.54 | russellb | Yourname``: there is no such thing as make upgrade |
| 05:04.57 | [TK]D-Fender | russellb, "PBX" isn't really the right term for use with *... |
| 05:05.00 | russellb | i have no idea what you are talking about |
| 05:05.11 | russellb | [TK]D-Fender: which is why it says and Tele... |
| 05:05.22 | Yourname`` | russellb: This guys said so -> http://users.pandora.be/Asterisk-PBX/UpdateAsterisk.htm |
| 05:05.39 | russellb | that's what you get for reading joe random blog |
| 05:05.43 | russellb | with authority of -47 |
| 05:05.43 | Yourname`` | I just want to update this SVN 1.4 I've been using to the latest and grreatest |
| 05:05.48 | [TK]D-Fender | russellb, Thats like listing a description of all the things you can build with a too, and not the fact thats its just a tool. |
| 05:05.53 | putnopvut | Yourname``, svn up |
| 05:05.59 | drmessano | Yourname``: Time for a saving throw |
| 05:06.18 | Yourname`` | Not svn though, just stable. |
| 05:06.52 | russellb | svn co http://svn.digium.com/svn/asterisk/branches/1.4 asterisk-1.4 |
| 05:06.52 | russellb | there |
| 05:06.53 | russellb | do that |
| 05:06.56 | drmessano | I'm confused |
| 05:07.08 | russellb | and run "make update" in there from now on |
| 05:07.22 | Yourname`` | Lemme try.. |
| 05:07.49 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.pa.comcast.net) |
| 05:08.07 | drmessano | SVN is kinda cool and creepy at the same time.. It's like being inside russellb's head at any given moment |
| 05:08.28 | Yourname`` | I dunno why, but SVN just scares me shitless. |
| 05:08.38 | drmessano | SVN is cool |
| 05:08.41 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.ct.comcast.net) |
| 05:08.45 | russellb | wtf |
| 05:08.48 | russellb | you just said how you want to run the latest |
| 05:08.50 | Yourname`` | Thankfully, last time it scared me.. russellb was there to help me out. |
| 05:08.51 | russellb | yet svn scares you? |
| 05:08.59 | fujin | git > svn |
| 05:09.13 | russellb | no scm > all |
| 05:09.14 | drmessano | s/svn/sharks/ ? |
| 05:09.15 | Yourname`` | Yeah, lol.. I wanna do it, but it's scary. Kinda like bunjee jumping. |
| 05:09.47 | russellb | soon we're going to open up svn for public commit access |
| 05:09.51 | russellb | screw reviewing patches |
| 05:10.08 | drmessano | lol |
| 05:10.21 | ZaVoid | bah i hate crappy wireless routers |
| 05:10.29 | ZaVoid | manx.. it kinda worked |
| 05:10.40 | ZaVoid | I can't get MacroIf to work... but GoToIf works |
| 05:10.44 | ZaVoid | so i guess that'll work |
| 05:11.38 | Yourname`` | I ws in /usr/src/asterisk and I did the svn co.. then there was asterisk1.4 in there. Without going into asterisk1.4, I did make update. And now I'm scared again.. |
| 05:11.53 | russellb | s |
| 05:11.58 | ZaVoid | sup russellb |
| 05:11.59 | russellb | Yourname``: mv asterisk-1.4 .. |
| 05:12.02 | putnopvut | outtolunc: shouldn't that be an app? |
| 05:12.03 | russellb | cd /usr/src/asterisk-1.4 |
| 05:12.20 | russellb | or something that just plugs into meetme, heh |
| 05:12.35 | russellb | ZaVoid: not much ... just messing around until i pass out |
| 05:12.42 | outtolunc | naw, they got plenty of those, this will be a 'voice' based bingo |
| 05:12.42 | ZaVoid | yeah same |
| 05:12.52 | Yourname`` | And then run make update? |
| 05:12.56 | drmessano | wget www.trixbox.org/download/haha/2.4.broken.iso may be a better idea |
| 05:13.02 | Yourname`` | Or go to make clean; make; make install? |
| 05:13.12 | putnopvut | drmessano, ooh broken is the best version! |
| 05:13.26 | putnopvut | It's radikewl! |
| 05:13.28 | drmessano | putnopvut: Broken is always the latest |
| 05:13.32 | outtolunc | but yeah i see your point |
| 05:14.15 | ZaVoid | so why would GoToIf work but MacroIf fail |
| 05:14.54 | ZaVoid | exten =>s,n,MacroIf("${arg4}"Â = "t")?macro-macronew,s,1) <-- fails... works if i change to GoToIf |
| 05:17.08 | [TK]D-Fender | ZaVoid, because you are not reading the apps instructions. |
| 05:17.15 | ZaVoid | i thought i did |
| 05:17.24 | [TK]D-Fender | ZaVoid, read again |
| 05:17.34 | ZaVoid | [Description] |
| 05:17.34 | ZaVoid | <PROTECTED> |
| 05:17.34 | ZaVoid | Executes macro defined in <macroname_a> if <expr> is true |
| 05:17.36 | ZaVoid | ok re-reading |
| 05:19.15 | putnopvut | I'm out. |
| 05:21.44 | Yourname`` | russellb: Got it! Thanks :) |
| 05:22.38 | ZaVoid | i get it now.. meh |
| 05:23.45 | *** part/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
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| 05:31.24 | bobbym | anyone using pyst to AGI programming ? |
| 05:32.04 | drmessano | HEH |
| 05:32.15 | drmessano | I guess I am getting my $12 out of my $12 BT adapter |
| 05:32.43 | drmessano | I got about 50 feet, thru a metal door |
| 05:33.44 | drmessano | I was hoping for about 20 more feet.. So if my phone got left in my truck, I wouldn't miss any calls |
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| 05:38.16 | obnauticus | Is anyone here really good with chan_mobile? |
| 05:38.39 | obnauticus | im having an old old problem with my stuff here is the debug |
| 05:38.42 | obnauticus | http://pastebin.ca/928654 |
| 05:40.57 | drmessano | http://bugs.digium.com/view.php?id=10824 |
| 05:41.12 | obnauticus | arrr err arr |
| 05:42.33 | obnauticus | drmessano mine does not even connect |
| 05:42.38 | obnauticus | it connects then disconnects(104) |
| 05:44.42 | drmessano | It may be incompatible |
| 05:45.01 | obnauticus | that would be hard gay |
| 05:45.10 | obnauticus | iirc my phone is compadible with headsets and stuff |
| 05:46.32 | drmessano | I don't know all the ins and outs, but I do know some devices don't work.. that may not be the case here, but it's worth checking the sparse documentation |
| 05:46.41 | drmessano | I think there's a list on the voip wiki |
| 05:47.16 | obnauticus | i never saw a list |
| 05:49.18 | obnauticus | I have this too drmessano: http://bugs.digium.com/file_download.php?file_id=17118&type=bug |
| 05:49.23 | *** join/#asterisk L|NUX (i=L_NUX@unaffiliated/lnux/x-10290) |
| 05:49.24 | drmessano | http://www.voip-info.org/wiki/view/chan_mobile |
| 05:51.24 | obnauticus | http://bugs.digium.com/view.php?id=11673 that's my bug report.... |
| 05:52.03 | jamesrdorn | hey guys I am trying to set a var for an inbound set of phone numbers to make my dialplan much easier to write. How can I call that var back during the dial string? Example: Set(ext=5555555555) then onto exten => ${ext},1,Answer(). I cant seem to get asterisk to view the var in the dial string in this manor. |
| 05:54.52 | jamesrdorn | nm, got it |
| 05:56.44 | lka | my FXO card can tell when a POTS line is busy, but can't seem to tell when there isn't a line connected at all |
| 05:56.51 | lka | does that sound right? |
| 06:00.30 | cootoa | is it possible to make an outgoing call from a phone through a trunk iax that has the second ipbx set to call on pstn? |
| 06:02.21 | cootoa | i have two ipbx with an iax trunk up and running |
| 06:03.09 | cootoa | one of the ipbx has the trunk and outbounds to call on pstn |
| 06:03.30 | cootoa | and the second one doesnt |
| 06:04.25 | cootoa | i want to use the iax trunk to reach that ipbx to call on pstn is it possible? |
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| 06:29.55 | jamesrdorn | is there a way to 'ping' a registered device to make sure it's able to take the call? My E51 has to be rebooted very often because it refuses to ring when a call comes in. Asterisk says "Ext 1000 is ringing" but, the phone just does not respond until reboot |
| 06:30.18 | jamesrdorn | while I know this is a bug w/ the phone, I need a way to have asterisk redirect the call when this happends |
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| 06:54.19 | ReDNeQ | sup sup |
| 06:55.08 | obnauticus | so drmessano it looks like right when this executes |
| 06:55.12 | obnauticus | ]2008-03-04 22:54:34] DEBUG[29113]: chan_mobile.c:980 rfcomm_write: rfcomm_write() (SPHA640) [AT+CIND? |
| 06:55.14 | obnauticus | the phone go meh |
| 06:55.57 | obnauticus | which woulkd be equal to this: RFCOMM(d): UIH: cr 0 dlci 8 pf 1 ilen 0 fcs 0x9c credits 1 |
| 06:57.05 | drmessano | ok |
| 06:57.11 | obnauticus | sauce?> |
| 07:00.18 | ReDNeQ | iax2? |
| 07:00.22 | ReDNeQ | ?iax2 howto |
| 07:00.24 | ReDNeQ | hrmm.. |
| 07:00.34 | obnauticus | ~iax2 |
| 07:00.35 | jbot | extra, extra, read all about it, iax2 is http://www.voip-info.org/wiki-IAX |
| 07:00.39 | obnauticus | Fail. |
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| 07:01.45 | ReDNeQ | obnauticus: thanks |
| 07:01.58 | obnauticus | you best be thankin |
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| 07:23.13 | SalocinTEN | hi, i have a problem with passing an incoming call to another voice menu, say like the first menu tells the user to press one, but when one is pressed, an engaged tone appears |
| 07:23.46 | SalocinTEN | ...instead of the relevant voice menu, which has already been set. |
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| 07:23.54 | SalocinTEN | how do i sort this problem out? |
| 07:24.34 | SalocinTEN | thanks. |
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| 07:56.08 | sergee | are there any rules in US which require ITSP to record calls and give it out by request from special services/courts? |
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| 08:00.12 | awk | hmm, anyone using 1.6 in production? |
| 08:00.22 | drmessano | That's scary |
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| 08:00.31 | awk | lol |
| 08:00.34 | awk | just wondering :) |
| 08:00.45 | drmessano | It's still Beta |
| 08:00.50 | awk | so is asterisk 1.4.18 |
| 08:00.53 | awk | but people still use it :) |
| 08:01.23 | drmessano | 1.4.18 is far from Beta.. 1.4 is finally becoming solid |
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| 08:08.51 | MatBoy | drmessano, 1.4.18 is very nice if you ask me |
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| 08:28.16 | MatBoy | a call will tkae about 8MB per call from mem as I'm right ? |
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| 08:30.13 | lnx | hi |
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| 08:39.02 | MatBoy | what is the perfomance difference between a Dual 1400Mhz or single P4 3Ghz ? It's strange to compare, but I found some examples for 3Ghz with about 150 calls... so I need some references form people if possible :) |
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| 08:43.43 | mchou | I'm a bit confused on how SIP is supposed to work. If I'm behind a NAT, can I have multiple instances of ATA/softphones registered to one provider on the same SIP port, w/o any undesirable "side effects?" |
| 08:43.55 | falco_toadfoot | is there a way to reset timeout of a second dial? |
| 08:44.18 | falco_toadfoot | first dial: dial(SIP/test,10,) |
| 08:44.25 | falco_toadfoot | phone sends 302 moved temporarily |
| 08:44.39 | falco_toadfoot | causing dial(SIP/someone-else) <--without timeout |
| 08:44.46 | falco_toadfoot | but it still times out at 10 seconds |
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| 09:13.49 | budol | hey |
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| 09:19.53 | falco_toadfoot | does anyone know how to reset the timer in a Dial(SIP/test,10,) ? |
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| 09:27.03 | kamisalami | hi all |
| 09:27.28 | kamisalami | can i ask a question? |
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| 09:28.34 | kamisalami | anyone here today? |
| 09:29.30 | falco_toadfoot | yes |
| 09:29.49 | kamisalami | ok, maybe you can help me.. who knows |
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| 09:30.27 | kamisalami | i am having this error message while loading misdn on my asterisk 1.4.10 (on ubuntu): |
| 09:30.33 | kamisalami | loader.c:373 load_dynamic_module: Module 'chan_misdn.so' did not register itself during load |
| 09:30.48 | kamisalami | i loaded the misdn modules |
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| 09:31.07 | kamisalami | configured them as i have them on another working box |
| 09:31.22 | kamisalami | but asterisk keeps complaining it can't load it |
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| 09:36.28 | ionix | hey, quick question: If I have a SIP provider in Japan, an asterisk server in Canada and a user in Japan, will the voice call be router through Canada or the asterisk will bridge the user and the sip provider? |
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| 09:38.26 | kamisalami | i think it will bridge if protocols differ, because it has to translate, but if protocols are the same, the voice will be send directly to the sip provider. |
| 09:38.31 | falco_toadfoot | kamisalami: don't know about that, sorry.. |
| 09:38.41 | kamisalami | no problem falco |
| 09:39.09 | kamisalami | i just keep on beating at the problem untill i solve it, but did not feel like reinventing the weel if i don't need to |
| 09:39.17 | falco_toadfoot | kamisalami: you wouldn't know of a way to reset the call timeout in a dial? |
| 09:39.27 | falco_toadfoot | this is the case: |
| 09:39.34 | falco_toadfoot | dial(SIP/test,10,) |
| 09:39.41 | falco_toadfoot | test sends a 302 redirect |
| 09:40.02 | falco_toadfoot | to a number which uses dial(SIP/other,20,) |
| 09:40.10 | falco_toadfoot | but the timeout of 10 seconds persists |
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| 09:41.29 | kamisalami | not sure about that, seems to work when i do it. maybe answer the call, then redirect? |
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| 09:42.34 | kamisalami | that would costs the calling party some phone fee though, as it is answered from that moment |
| 09:44.35 | kamisalami | actually, answer, then dial. that's what i meant |
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| 09:45.25 | kamisalami | i find it strange that it does not reset the counter, but that probably has something to do with the redirect coming from the phone, not asterisk itselve |
| 09:46.13 | kamisalami | actually, the more i think about it, the more sense it makes |
| 09:46.45 | lnx | best way to get ${VARIABLE} with AGI perl script is exten => s-NOANSWER,1,AGI(logscript.pl,${DIALSTATUS}) and use ARGV[] :))) get_var is hard pain :) |
| 09:47.12 | *** join/#asterisk BeeBuu (n=beebuu@59.38.97.193) |
| 09:47.23 | BeeBuu | hello,all |
| 09:47.33 | BeeBuu | anyone using the PAGE() command? |
| 09:47.39 | kamisalami | can't you configure the phone answering the sip/test to not redirect and do the redirect on asterisk itselve? |
| 09:47.42 | kamisalami | no , sorry |
| 09:47.49 | kamisalami | beebuu |
| 09:47.57 | falco_toadfoot | kamisalami: thanks for the info |
| 09:48.26 | falco_toadfoot | kamisalami: yes, I could, but then there would be no information on the phone display to show it's redirecting calls |
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| 09:54.50 | kamisalami | true |
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| 10:11.34 | kamisalami | i have 2 things loading misdn modules: |
| 10:11.43 | kamisalami | misdn-init |
| 10:11.53 | kamisalami | mISDN |
| 10:12.15 | kamisalami | both are in /etc/init.d/ |
| 10:12.19 | kamisalami | is that normal? |
| 10:12.27 | kamisalami | or should i use only one of them? |
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| 10:32.17 | angryuser[A] | <kamisalami> dont touch if it works |
| 10:32.22 | angryuser[A] | :) |
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| 10:36.14 | kamisalami | it does not |
| 10:36.28 | kamisalami | [Mar 5 11:28:54] NOTICE[7808]: loader.c:831 load_modules: 159 modules will be loaded. |
| 10:36.28 | kamisalami | [Mar 5 11:28:54] WARNING[7808]: loader.c:373 load_dynamic_module: Module 'chan_misdn.so' did not register itself during load |
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| 10:44.03 | tzafrir | kamisalami, that error is of an Asterisk module, not of the Linux kernel module |
| 10:44.21 | kamisalami | i know |
| 10:44.50 | kamisalami | just don't know why it is giving me that |
| 10:45.12 | kamisalami | tried enabling core debug , but did not give any more info on it |
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| 10:48.06 | *** join/#asterisk Daejeo (n=chatzill@211.177.189.62) |
| 10:48.37 | Daejeo | anyone from Europe? |
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| 10:49.09 | Daejeo | whats the terminating impedance for fxs port? |
| 10:49.31 | exvito | asterisk 1.4 seems to have two modes for the SayXXX applications... OLD mode and NEW mode, based on say.conf... it seems to startup in OLD mode... after peeking at the source i found no obvious way of starting it up in NEW mode... any hints ? |
| 10:50.39 | tzafrir | Daejeo, hmmm... why do you ask? |
| 10:51.36 | tzafrir | Daejeo, from what country in europe? |
| 10:51.37 | Daejeo | i have to install PAP2 ata |
| 10:51.37 | Daejeo | england |
| 10:52.13 | tzafrir | FXS port? to connect a phone? or an FXO port , to connect to the PSTN? |
| 10:52.17 | zeeesh | i hv registered my 2 sip peers peer 1000 and peer 200 ... both can call each other and dialout .... after an hour when i tried to dial out they r working but when they try to call each other i got msg |
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| 10:53.39 | Daejeo | 370+620||310nF |
| 10:54.13 | zeeesh | i hv registered my 2 sip peers peer 1000 and peer 200 ... both can call each other and dialout .... after an hour when i tried to dial out they r working but when they try to call each other i got msg "-- SIP/100-095e6b80 is circuit-busy" "-- Executing Hangup("SIP/100-0960b158", "")"? |
| 10:54.40 | Daejeo | tzafrir: ata does not have any fxo port |
| 10:55.05 | tzafrir | Daejeo, so it only depends on your phone |
| 10:55.31 | Daejeo | my phone is made in uk |
| 10:55.35 | Daejeo | so i asked |
| 10:55.55 | *** join/#asterisk g0mb0 (n=root@external.micom.mng.net) |
| 10:56.04 | Daejeo | i need to set the impedance according that |
| 10:56.33 | awk | tzafrir: what complications would you get running say chan_iax2.so from 1.4.18 on 1.4.17 ? |
| 10:56.50 | tzafrir | awk, no idea |
| 10:56.56 | Daejeo | Selects the impedance of the analog telephone connected to the Phone port. |
| 10:56.57 | Daejeo | The following information may be useful for end user configuration: |
| 10:56.59 | Daejeo | 600Ohm – North America |
| 10:57.00 | Daejeo | 270+750Ohm || 150nF – Most of Europe |
| 10:57.02 | Daejeo | 220+820Ohm || 120nF – Australia, New Zealand |
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| 10:57.04 | Daejeo | 220+820Ohm || 115nF – Austria, Bulgaria, Germany, Slovakia, South Africa |
| 10:57.05 | Daejeo | 370+620Ohm || 310nF – UK, India |
| 10:57.06 | awk | arg |
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| 10:57.50 | Daejeo | tazfir |
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| 10:58.37 | Daejeo | ok? |
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| 11:12.54 | cootoa | hello |
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| 11:13.26 | cootoa | someone feeling to give me a hand |
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| 11:15.13 | cootoa | looking some hint to make outgoing call from a trixbox link to a second one with an iax trunk that has the outbound routes to make outgoing calls |
| 11:15.19 | ice_croft | why do i need "sip trunk" instead of direct sip? |
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| 11:16.56 | cootoa | its because they want an ipbx on each site to make local call |
| 11:16.56 | awk | ice_croft ? |
| 11:17.01 | cootoa | in the site |
| 11:17.36 | ice_croft | awk> ? |
| 11:18.11 | cootoa | from each site they want one ipbx setup and to make the outgoin calls they want to go through iax trunk to reach the ipbx that has the route to make out goin calls |
| 11:18.32 | cootoa | so they can control the outgoin calls |
| 11:20.19 | cootoa | mainly they wan an iax trunk to make inter-site call |
| 11:20.33 | cootoa | that i succeed to put up |
| 11:21.30 | *** join/#asterisk simbol76ss (n=simbol@host228-198-dynamic.49-82-r.retail.telecomitalia.it) |
| 11:21.38 | cootoa | then they want the outgoing calls to go though iax trunk to reach the ipbx that can redirect to outgoing calls |
| 11:22.57 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
| 11:23.24 | dominic1 | Hello guys, I am using asterisk 1.4.2 and I need the remove extension command to reset a hint |
| 11:23.53 | dominic1 | I am not able to find the command. Will I have to adjust a setting to use this command |
| 11:24.10 | dominic1 | on another system with asterisk 1.4.17 it's not a problem to use this command |
| 11:28.03 | *** part/#asterisk exvito (n=exvito@195.245.132.93) |
| 11:29.19 | dominic1 | Can anybody help me? |
| 11:30.57 | tzafrir | ~ask |
| 11:30.58 | jbot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there, just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 11:31.07 | tzafrir | ah, you did |
| 11:31.39 | tzafrir | dominic1, dialplan remove ... |
| 11:32.53 | lnx | is possibble to run a CLI comand example sip show registry from AGI script? |
| 11:33.00 | *** join/#asterisk qdk (n=qdk@195.242.194.42) |
| 11:37.36 | cootoa | i have 2 ipbx one can call the outside the second one cant.the 2 ipbx are link with an iax trunk, how can i configurate the second ipbx(cant call outside) to use the route of the ipbx that can call outside via the iax trunk |
| 11:37.40 | cootoa | thanks |
| 11:41.45 | mattman99 | coota, you need to give the 2nd PBX a trunk route to the first PBX |
| 11:43.14 | mattman99 | coota, somthing like "x." then trunk should be 1st pbx IAX trunk |
| 11:44.12 | *** join/#asterisk ArashHemmat (n=ArashHem@91.184.77.35) |
| 11:44.36 | cootoa | can you give me more details? |
| 11:45.12 | mattman99 | are u using CLI or freepbx etc? |
| 11:45.35 | *** join/#asterisk ice_croft (n=nolan@85.172.5.106) |
| 11:45.50 | cootoa | freepbx |
| 11:47.18 | mattman99 | so server1 = voip providers and server2 = remote site with IAX trunk connecting them correct? |
| 11:47.52 | cootoa | yes |
| 11:49.11 | kamisalami | anyone here who knows what is wrong when kernel modules are loaded, but i still get: [Mar 5 11:28:54] WARNING[7808]: loader.c:373 load_dynamic_module: Module 'chan_misdn.so' did not register itself during load |
| 11:49.26 | kamisalami | sorry for the bold... did not do that on purpose |
| 11:50.46 | cootoa | can you guide me through the steps ? |
| 11:52.22 | *** join/#asterisk loproc (n=no@cpe.atm2-0-90374.0x503f7912.rdnxx6.customer.tele.dk) |
| 11:52.57 | loproc | Hey. I need some help on configuring Asterisk to connect to a sip provider (in order to be able to recieve PSTN calls) |
| 11:53.43 | *** join/#asterisk chronos (n=fuckyou@189.7.85.214) |
| 11:53.50 | chronos | hello for everybody |
| 11:54.14 | loproc | hello chronos |
| 11:55.50 | mattman99 | cootoa, just need to add x. in your trunk to server1 |
| 11:56.04 | mattman99 | sorry, outgoing dialplan |
| 11:56.40 | chronos | I wanting to make a callback like as in www.justvoip.com. The webcallback call two non voip numbers. How I do it? use a normal callback script that call a SIP and in specific context I call first and after call second number ? |
| 11:57.36 | *** join/#asterisk chronos (n=fuckyou@189.7.85.214) |
| 11:57.55 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 11:57.56 | cootoa | so on server 1 in the iax trunk that link the server 2 ineed to add x. in the dialplan? |
| 11:58.30 | chronos | (I disconnected, not see replies in last seconds) |
| 11:58.50 | flush | yo |
| 11:58.56 | flush | i got a problem |
| 11:59.16 | chronos | hum, and if is how I say, calling all via context, how I can call two numbers and connect both via asterisk ? |
| 11:59.54 | flush | i have asterisk installed with a tdm400p, i configured it using http://downloads.oreilly.com/books/9780596510480.pdf book and now i press 500, i get the repeat thing, how do i place call to the outside world?? also, when i call home with my cell phone, asterisk doesnt ring the phone plugged in the FXS module |
| 11:59.56 | flush | what gives? |
| 11:59.57 | chronos | the dial runs in background? |
| 12:00.17 | *** join/#asterisk axscode (i=acausing@203.160.164.254) |
| 12:00.39 | axscode | hi guys,, anyone have a sip client? i just wanted to test my box |
| 12:00.43 | axscode | pm please.. thanks |
| 12:01.29 | *** join/#asterisk NirS (n=NirS@87.68.17.141.cable.012.net.il) |
| 12:02.18 | cootoa | i forgot to mention phone link to server 1 dial either 8 or 9 before dialing the phone number |
| 12:03.50 | Der-Tim | hi there |
| 12:03.57 | Der-Tim | i got a problem calling from one trixbox to another via a vanilla asterisk "hub"... |
| 12:04.02 | Der-Tim | at the remote trixbox, i get an error with iax2 debug which says, that no authority is found... |
| 12:04.06 | Der-Tim | the local trixbox calls out to the hub, the hub sends my call to the remote one (with correct caller id and everything correct), but the remote one rejects the call with the given error... |
| 12:04.24 | Der-Tim | any ideas? |
| 12:04.42 | Der-Tim | there is a trunk from each trixbox to the hub and also back... |
| 12:05.48 | *** join/#asterisk SteveTotaro (n=root@pool-71-179-127-222.bltmmd.east.verizon.net) |
| 12:06.15 | axscode | hi guys... anyone knows any console based softphone..? |
| 12:07.38 | *** join/#asterisk ruied (n=ruied@pal-213-228-184-21.netvisao.pt) |
| 12:09.37 | *** join/#asterisk keulin (n=cray@ifth-pdcr2.infotheme.fr) |
| 12:10.19 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 12:12.42 | *** join/#asterisk MmixX (i=mmixx@202.124.138.69) |
| 12:18.09 | lnx | axscode: - |
| 12:18.22 | cootoa | please help |
| 12:18.29 | axscode | lnx... |
| 12:18.50 | lnx | axscode: no such client |
| 12:19.02 | axscode | ok there is pjsip |
| 12:19.12 | axscode | but nevermind |
| 12:21.08 | lnx | axscode: indeed, cool |
| 12:21.46 | axscode | :) |
| 12:22.46 | loproc | Anyone with an idea of why I can only recieve two concurrent calls from another SIP provider? I've got the "register" line in my sip.conf forwarding to an internal extension but the third caller simply gets a dialtone. The SIP provider says they should have no problem forwarding more than two simultaneous calls... |
| 12:23.35 | *** join/#asterisk telenieko (i=marc@167.Red-80-35-144.staticIP.rima-tde.net) |
| 12:24.53 | telenieko | Hi ppl. I have a SPA942 (behind nat) that works fine except when transfering, the REFER message it sends has "Via" header with the LAN IP, not the WAN one. any other message sent by the phone has the correct Via header. How can I make Asterisk accept the wrong Via? (It currently answers with a 603 no dialog error..) |
| 12:25.14 | *** join/#asterisk penguinFunk (n=penguin@unaffiliated/penguinfunk) |
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| 12:57.03 | kamisalami | i am having troubles loading chan_misdn.so : |
| 12:57.04 | kamisalami | .[Mar 5 13:56:08] WARNING[8053]: loader.c:373 load_dynamic_module: Module 'chan_misdn.so' did not register itself during load |
| 12:58.19 | kamisalami | after that i get also: |
| 12:58.19 | kamisalami | [Mar 5 13:57:43] WARNING[8085]: loader.c:373 load_dynamic_module: Module 'chan_misdn.so' did not register itself during load |
| 12:58.19 | kamisalami | [Mar 5 13:57:43] WARNING[8085]: loader.c:637 load_resource: Module 'chan_misdn.so' could not be loaded. |
| 12:58.38 | kamisalami | so it seams to attempt to load it twice |
| 13:03.20 | *** join/#asterisk ZaVoid (n=zavoid@66-95-182-90.client.dsl.net) |
| 13:07.20 | *** join/#asterisk BipBip (n=BipBip@194.65.5.235) |
| 13:08.32 | *** join/#asterisk ManxPower (n=manxpowe@ip72-204-168-61.no.no.cox.net) |
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| 13:15.11 | cr0n | during a call, how does one check which codec is been used, just to make sure its using what is specified in the config? |
| 13:16.23 | ManxPower | sip show channels |
| 13:16.57 | cr0n | excellent, thanks. |
| 13:23.19 | *** join/#asterisk ruied (n=ruied@pal-213-228-184-21.netvisao.pt) |
| 13:26.50 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.117.5) |
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| 13:39.36 | cr0n | any simple providers out there allowing free routing to 1800 numbers? |
| 13:39.55 | *** join/#asterisk l2cache (n=l2cache@64.128.254.98) |
| 13:40.35 | *** join/#asterisk lesouvage (n=lesouvag@62.140.137.125) |
| 13:42.05 | actros1840 | did you try SiPbroker ? |
| 13:42.16 | dominic1 | dialplan remove is not available |
| 13:42.29 | *** join/#asterisk simbol76ss (n=simbol@ip-212-26.sn1.eutelia.it) |
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| 14:01.12 | lesouvage | . |
| 14:04.38 | Der-Tim | hi there |
| 14:04.40 | Der-Tim | anyone alive? |
| 14:05.44 | ManxPower | We are all dead. You are on #asterisk-zombies. |
| 14:05.54 | ManxPower | How can we help WANT TO EAT BRAINS you |
| 14:06.08 | lesouvage | der-tim: I am but nothing to ask |
| 14:06.15 | moemoe | BRAAAAAAAINS! BRAAAAAAAINS! |
| 14:06.30 | coppice | its shaun of the dead 2 |
| 14:07.24 | *** join/#asterisk ddunavant (n=David@pool-71-178-109-242.washdc.east.verizon.net) |
| 14:07.29 | *** join/#asterisk RoyK (n=roy@box36.fortel.no) |
| 14:07.50 | kamisalami | may i? |
| 14:07.53 | kamisalami | may i? |
| 14:07.56 | kamisalami | <kamisalami> i am having troubles loading chan_misdn.so : |
| 14:07.56 | kamisalami | <kamisalami> .[Mar 5 13:56:08] WARNING[8053]: loader.c:373 load_dynamic_module: Module 'chan_misdn.so' did not register itself during load |
| 14:07.56 | kamisalami | <kamisalami> after that i get also: |
| 14:07.56 | kamisalami | <kamisalami> [Mar 5 13:57:43] WARNING[8085]: loader.c:373 load_dynamic_module: Module 'chan_misdn.so' did not register itself during load |
| 14:07.56 | kamisalami | <kamisalami> [Mar 5 13:57:43] WARNING[8085]: loader.c:637 load_resource: Module 'chan_misdn.so' could not be loaded. |
| 14:07.59 | kamisalami | <kamisalami> so it seams to attempt to load it twice |
| 14:08.18 | kamisalami | oh my brains |
| 14:08.20 | kamisalami | bummer |
| 14:08.24 | kamisalami | i don't have any |
| 14:09.14 | kamisalami | that's why i ask my q's here |
| 14:10.21 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:13.01 | Der-Tim | anyone knows what will happen if an asterisk server is configured to put all dial patterns on an zap trunk and a user internally dials the extension of an sip-phone eg? |
| 14:13.39 | denon | zap will dial whatever you tell it to .. |
| 14:13.41 | ManxPower | Der-Tim: dialpatterns with SIP don't work the same as with Zap. |
| 14:13.44 | denon | the pstn may not know what to do with it |
| 14:13.55 | denon | oh, maybe I misunderstood him |
| 14:14.02 | ManxPower | If the dialed digits pass the SIP Phones INTERNAL dialplan, then of course asterisk will route the call however extensions.conf tells it. |
| 14:14.34 | ManxPower | kamisalami: that looks like you have a chan_misdn from 1.2 and are trying to use it in 1.4. That won't work. |
| 14:15.05 | Der-Tim | ManxPower: well, i assigned "." (in trixbox) to be routed via zap... what will happen, if a local user dials eg extension 100 which is on the same pbx as he is? |
| 14:15.21 | ManxPower | ~trixbox. |
| 14:15.22 | jbot | [~trixbox] trixbox is a full linux distro that includes FreePBX and other 3rd party add-ons. It is all this extra stuff which makes trixbox VERY difficult to support, and is not supported in #asterisk. Try asking in #trixbox or on their forums & wiki at http://www.trixbox.org |
| 14:15.29 | kamisalami | Manx: Hmz.... i just used the packages from ubuntu |
| 14:15.32 | ManxPower | I thought I smelled the stench of a GUI. |
| 14:15.48 | *** join/#asterisk tobias (n=tobias@cpe-066-026-085-055.nc.res.rr.com) |
| 14:15.49 | ManxPower | kamisalami: We can't help you if you don't compile from source. |
| 14:16.08 | kamisalami | ok, well, i will consider it then |
| 14:16.16 | denon | or more importantly, start with blank or sample configs |
| 14:16.18 | ManxPower | Der-Tim: We can't even *begin* to help you. |
| 14:16.39 | tzafrir | "blank" and "sample config" are two different things |
| 14:16.42 | ManxPower | kamisalami: do you need MISDN? |
| 14:16.47 | denon | tzafrir: well, asterisk-sample |
| 14:16.47 | tzafrir | kamisalami, what exactly is the issue? |
| 14:16.50 | kamisalami | ehm.... you did begin, guess i am lucky :) |
| 14:16.52 | ManxPower | neither will work if the module is broken. |
| 14:17.00 | kamisalami | manx: yeah, to talk to my isdn card |
| 14:17.12 | tzafrir | asterisk-sample is a sample. And quite confusing as to what is the default |
| 14:17.13 | ManxPower | kamisalami: compile from source or complain to the person that made the package |
| 14:17.21 | kamisalami | i had same setup on a custom build machine |
| 14:17.25 | tzafrir | kamisalami, what version of Ubuntu? |
| 14:17.33 | denon | tzafrir: sure, but at least we can have him rm -rf those then ;) |
| 14:17.42 | tzafrir | At least some versions of Ubuntu are shipped with a broken version of misdn |
| 14:17.49 | kamisalami | ubuntu server, 7.1 if i am right |
| 14:17.50 | *** join/#asterisk seanbright (i=seanbrig@65.207.74.18) |
| 14:18.00 | tzafrir | 7.10, actually |
| 14:18.10 | kamisalami | yeah |
| 14:18.13 | kamisalami | 7.10 |
| 14:18.19 | tzafrir | Otherwise you could think it is not newer than 7.4 |
| 14:18.34 | kamisalami | sry for that, failing memory :) |
| 14:18.39 | tzafrir | What version of misdn does it have? |
| 14:18.53 | tzafrir | What misdn card do you have? |
| 14:18.54 | kamisalami | i custom build it from newest sources |
| 14:19.03 | *** join/#asterisk Datax (n=Datax@glou.nurvnet.org) |
| 14:19.19 | *** join/#asterisk jetlagmk2 (i=jetlag@70.17.48.245) |
| 14:19.29 | kamisalami | mISDN-1_1_7_2 |
| 14:19.51 | *** join/#asterisk jmesquita (n=jmesquit@200.170.114.149) |
| 14:20.07 | ManxPower | kamisalami: and what version of asterisk does that say it's for? |
| 14:20.16 | tzafrir | kamisalami, hmm... so how do you expect to use Asterisk from packages if you use mISDN from source? |
| 14:20.29 | tzafrir | are the two compatible enough?> |
| 14:20.36 | kamisalami | tzxafrir: i did not know i could not |
| 14:20.48 | *** join/#asterisk phillipk (n=phillipk@fw.datafax.net) |
| 14:20.51 | tzafrir | I'm not sure how stable the interface is |
| 14:20.56 | kamisalami | i failed to find any supporting misdn packages in ubuntu |
| 14:21.01 | tzafrir | I really don't know much about mISDN |
| 14:21.05 | JayTee52 | anyone here used a Redfone foneBridge2-EC? |
| 14:21.18 | tzafrir | kamisalami, ubuntu here mostly follows Debian |
| 14:21.24 | kamisalami | i use it, but on a custom build server, there it works fine |
| 14:21.26 | tzafrir | what card do you have |
| 14:21.38 | kamisalami | moment, have to check |
| 14:22.05 | kamisalami | Network controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface |
| 14:22.15 | kamisalami | output taken from lspci |
| 14:22.34 | tzafrir | kamisalami, hmmm.... is that an ISDN card, or a Digium analog card? |
| 14:23.09 | kamisalami | your right, pasted wrong one |
| 14:23.10 | kamisalami | Network controller: Cologne Chip Designs GmbH ISDN network controller [HFC-PCI] (rev 02) |
| 14:23.37 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 14:23.41 | tzafrir | well, that one is also supported by bristuff, if you give up on misdn |
| 14:23.43 | *** join/#asterisk Entr4nced (n=IMG001@68-247-96-65.area3.spcsdns.net) |
| 14:24.04 | kamisalami | i'll keep it in mind |
| 14:24.09 | ManxPower | Me to a consultant: "Some people at Stirling do store important documents in their trash. We think it's a genetic anomaly." |
| 14:24.09 | kamisalami | thnx for the help |
| 14:24.25 | kamisalami | hehehe |
| 14:24.40 | kamisalami | i have those ppl here also |
| 14:24.59 | tzafrir | the packages of asterisk / zaptel you have include bristuff patches already |
| 14:25.18 | kamisalami | ok, i'll check if i can use bristuff instead |
| 14:25.42 | tzafrir | you *can* use both |
| 14:25.53 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 14:26.28 | kamisalami | i want to use it with hylafax |
| 14:26.46 | kamisalami | i merely used misdn because it is on my other box as well, and there it is working fine |
| 14:27.09 | kamisalami | but now i want to set up a system using a minimum of user built sources |
| 14:27.22 | kamisalami | so i think i will go with that if hylafax can use it |
| 14:32.02 | kamisalami | where did you get the info about the bristuff patches that are included from? |
| 14:32.51 | gego | Hello, can anyone please help me with the following problem: I'd like to watch the DEVSTATE of 20 GXPs to avoid calling a busy phone. |
| 14:33.35 | gego | I can see if there's already an active channel with GROUP() and GROUP_COUNT() if they are called directly or initiated the call. |
| 14:34.16 | *** join/#asterisk Juggie (i=Juggie@CPE001601df17fb-CM001a6682d7b6.cpe.net.cable.rogers.com) |
| 14:34.26 | gego | My problem is to determine, who actually accepted a call from a multidial. |
| 14:35.00 | gego | I'm using * 1.4 and DEVSTATE is introduced in 1.6 i think? |
| 14:36.21 | *** join/#asterisk putnopvut (n=putnopvu@216.207.245.1) |
| 14:36.21 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:37.34 | *** join/#asterisk stansmith (n=stansmit@adsl-68-252-62-180.dsl.wotnoh.ameritech.net) |
| 14:37.35 | *** join/#asterisk jameswf-home (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 14:37.43 | stansmith | ~hi all |
| 14:37.44 | jbot | Many greetings, all, most strange traveller, to this IRCdom of plenty. |
| 14:37.52 | [TK]D-Fender | gego: "core show application chanisavail" <--- |
| 14:38.11 | jameswf | ~42 |
| 14:38.12 | jbot | somebody said 42 was the answer to life the universe and everything, see also http://en.wikipedia.org/wiki/the_answer_to_life,_the_universe,_and_everything |
| 14:38.20 | coppice | jbot is truly ircsome |
| 14:38.40 | stansmith | jbot represents everything that is great about america |
| 14:38.41 | jbot | okay, stansmith |
| 14:39.00 | Katty | mew. |
| 14:39.00 | coppice | jbot ~represents |
| 14:39.01 | jameswf | ummm |
| 14:39.04 | [TK]D-Fender | ~job |
| 14:39.05 | jbot | Then Cladhaire arose, and rent his mantle, and shaved his head, and fell down upon the ground, and worshipped. And said, Naked came I out of my mother's womb, and naked shall I return thither: the Blizzard gave, and the Blizzard hath taken away; blessed be the name of the Slouken. -- Cide 1: 20-21 |
| 14:39.24 | jameswf | ~jbot |
| 14:39.25 | jbot | it has been said that jbot is a hack!, or known to have only said one useful thing. a tool, or dating the fembots, or [TK]D-Fender's b*tch, or suck |
| 14:39.29 | kamisalami | <tzafrir: where did you find the info about the ubuntu packages including the bristuff patches? |
| 14:39.56 | coppice | jbot t.38 is a pain in the ass |
| 14:39.57 | jbot | ...but t.38 is already something else... |
| 14:40.11 | [TK]D-Fender | ~t.38 |
| 14:40.12 | jbot | it has been said that t.38 is see t38 |
| 14:40.17 | [TK]D-Fender | ~t38 |
| 14:40.18 | jbot | from memory, t38 is see http://www.brooktrout.com/whitepapers/pdf/fax_over_ip.pdf for a decent overview of how it all works, no, it's not ready yet, we'll let you know. a really lousy spec. a lightweight fighter, also known as the Talon |
| 14:40.24 | coppice | jbot jbot is a pain in the ass |
| 14:40.25 | jbot | ...but jbot is already something else... |
| 14:40.51 | seanbright | jbot jbot is also a pain in the ass |
| 14:40.52 | jbot | seanbright: okay |
| 14:40.56 | coppice | ~t.37 |
| 14:41.26 | coppice | jbot t.37 is what t.38 wants to be when it grows up |
| 14:41.27 | jbot | coppice: okay |
| 14:41.45 | anonymouz666 | the link for t.38 is broken |
| 14:42.17 | coppice | that's because brooktrout begat cantata, who begat dialogic |
| 14:42.29 | jameswf | ~canada |
| 14:42.29 | jbot | We're sorry we called your president a moron, I mean he IS a moron but it wasn't nice of us to point it out. besides it's not like you voted him into office. |
| 14:42.35 | *** join/#asterisk dhill (i=dhill@fog.mindcry.org) |
| 14:43.11 | coppice | there should be a dialogic link for that T.38 black and white paper |
| 14:43.24 | dhill | When I VoIP user makes an outbound call to another VoIP user on the same asterisk system, asterisk only logs the outbound call in Master.csv, but not an additional inbound call for the called party. Is this intentional? |
| 14:44.10 | jameswf | ~bush |
| 14:44.11 | jbot | ooooh Bush me no likey bring on the tile baby... wanna borrow my razor.... |
| 14:44.36 | jameswf | ~gop |
| 14:44.46 | jameswf | doh |
| 14:45.53 | [TK]D-Fender | dhill: Perfectly normal. |
| 14:45.55 | JayTee52 | so has anyone in here even heard of the Redfone foneBridge2? Good or bad? |
| 14:46.10 | [TK]D-Fender | JayTee52: Bleh |
| 14:46.18 | Katty | mew? |
| 14:46.19 | JayTee52 | Bleh? why so? |
| 14:46.22 | [TK]D-Fender | JayTee52: Nobody gives a shit about TDMoE |
| 14:46.30 | [TK]D-Fender | JayTee52: Crappy EC, etc |
| 14:46.38 | jameswf | ~redphone |
| 14:46.42 | stansmith | jbot, common sense isnt that common |
| 14:46.43 | jameswf | ~redfone |
| 14:46.53 | Katty | MEW?! |
| 14:46.53 | jameswf | bah |
| 14:47.02 | stansmith | curse u jbot! |
| 14:47.03 | Katty | oh. wait. english. |
| 14:47.04 | Katty | HI! |
| 14:47.22 | kamisalami | routing call's to a bristuff isdn supported card, connected to an mobile gateway, will that work |
| 14:47.28 | jameswf | I is fluent in techonise |
| 14:47.29 | kamisalami | ? |
| 14:48.33 | dhill | [TK]D-Fender: anyway to get asterisk to log both? |
| 14:48.45 | Katty | jameswf: that's not something to be proud of. |
| 14:48.52 | jameswf | heh |
| 14:49.02 | coppice | jameswf: kneeing him in the nuts will do that in seconds |
| 14:49.03 | Katty | jameswf: work on patience instead (= |
| 14:49.08 | Katty | jameswf: much more productive in life. |
| 14:49.18 | [TK]D-Fender | dhill: You shouldn't. but you could loop everything back in through a local channel.... |
| 14:50.11 | coppice | jameswf: they have a comedy show on bangalore TV making fun of people like you |
| 14:50.24 | jameswf | oh allo muh name is john. I am fron can-sas. the weather is verry much nice here in can-sas |
| 14:51.00 | nixguy | the default messages in asterisk, theyr'e made with some voice synth application is it free? |
| 14:51.17 | coppice | oh, can-sas is just down the road from the new bangalore airport that opens next month |
| 14:51.17 | Katty | [TK]D-Fender: GOOD MORNING SNOBBY |
| 14:51.26 | [TK]D-Fender | Katty: Mew. |
| 14:51.30 | Katty | [TK]D-Fender: that's more like it. ^_^ |
| 14:51.36 | jameswf | allison isnt a voice synth ap... i dont think I shal email and ask her |
| 14:51.38 | stansmith | nixguy: i think they had allison smith record those herself |
| 14:51.54 | nixguy | who is allison smith? |
| 14:52.01 | stansmith | sex goddess |
| 14:52.04 | nixguy | so i have to record new ones with my own voice? |
| 14:52.10 | nixguy | that would scare all customers away |
| 14:52.16 | stansmith | cepstral has a TTS engine with the allison voice |
| 14:52.16 | dhill | [TK]D-Fender: it is for billing purposes. We want the call to show up on both customers bill. We show inbound calls as well, we just dont charge.. |
| 14:53.17 | Qwell | jameswf: you could always hire her.. |
| 14:53.22 | *** join/#asterisk outtolunc (n=me@c-67-174-216-60.hsd1.ca.comcast.net) |
| 14:53.26 | jameswf | nah I am to cheap... |
| 14:53.51 | jameswf | I have a $10 Dennys gift card lol |
| 14:54.42 | tzafrir | kamisalami, I know that the ubuntu packages are basically copies of the respective Debian packages |
| 14:54.55 | tzafrir | And I do know the Debian packages :-) |
| 14:56.23 | stansmith | exten=>h,1,AGI(my_script.vb) <---- is calling AGI as first priority on hangup some kind of logical fallacy? |
| 14:56.26 | gego | [TK]D-Fender: ChanIsAvail seems to lead in the right direction, but may also indicate an internal call. Thanks, anyway. |
| 15:00.03 | tzafrir | kamisalami, also generally try: http://packages.ubuntu.com/<packagename> |
| 15:00.25 | tzafrir | Look at the specific package for each package. There's also a link there to the changelog |
| 15:00.35 | *** join/#asterisk brunner (n=chris@66.35.172.123) |
| 15:01.05 | brunner | What kind of hardware would I need to support 64 phone calls at a time? |
| 15:01.20 | Qwell | brunner: not much |
| 15:01.32 | stansmith | !X11 |
| 15:01.38 | brunner | Qwell: like four computers? |
| 15:01.41 | brunner | eight? |
| 15:01.47 | Qwell | better make it 12 |
| 15:01.55 | *** join/#asterisk abaci (n=IceChat7@ool-4b7fc532.static.optonline.net) |
| 15:01.58 | Qwell | no, really, 1 1ghz box would be more than enough for 64 calls |
| 15:02.02 | brunner | okay, and a specialized PCI card or something? |
| 15:02.04 | brunner | oih |
| 15:02.37 | Qwell | brunner: depends on the types of calls you'll be doing |
| 15:02.52 | brunner | Qwell: interactive touch-tone menus |
| 15:03.06 | Qwell | I mean, the tech you use. analog/T1 vs voip |
| 15:03.15 | brunner | oh |
| 15:03.18 | brunner | say it's voip |
| 15:03.34 | Qwell | then no pci cards or anything like that |
| 15:03.39 | *** join/#asterisk outtolunc (n=me@c-67-174-216-60.hsd1.ca.comcast.net) |
| 15:03.41 | brunner | wow |
| 15:03.55 | brunner | and a 1Ghz machine with no special hardware can reliably handle 64 calls at once? |
| 15:04.00 | stansmith | 09:51 <@Qwell> no, really, 1 1ghz box would be more than enough for 64 calls |
| 15:04.01 | Qwell | brunner: more or less |
| 15:04.03 | stansmith | wtf |
| 15:04.05 | kamisalami | thnx tzafrir, i appreciate your help |
| 15:04.23 | stansmith | sorry for flood |
| 15:04.23 | Qwell | brunner: I mean, you're going to want more than 1ghz, but it should handle it easily |
| 15:04.36 | Qwell | (assuming nothing else on the box, no transcoding of audio, etc, etc) |
| 15:04.38 | brunner | okay, so a nice server then |
| 15:05.01 | brunner | but it could play mp3's or some audio file for the menus? |
| 15:05.05 | Qwell | depends on your definition of "nice".. anything I would consider "nice" would be overkill :) |
| 15:05.08 | Qwell | yep |
| 15:05.26 | Katty | hi Qwell |
| 15:05.31 | brunner | okay, what kind of bandwidth would be required to handle 64 calls at a time? |
| 15:05.58 | Qwell | brunner: depends on the codec. worst case scenario is g711, at 64k/s, plus overhead (which ends up at around 85k/s) |
| 15:06.21 | Qwell | erm, per call |
| 15:06.56 | brunner | does GSM sound bad to most people? |
| 15:07.21 | Qwell | not most. |
| 15:07.38 | Qwell | it's "okay".. definitely not great |
| 15:07.52 | *** join/#asterisk shido6 (n=shido6@204.126.120.132) |
| 15:07.59 | Qwell | I think with overhead, gsm is around...30k/s? |
| 15:08.05 | brunner | what's a commonly-supported (by voip providers) low bandwidth codec? |
| 15:08.09 | Qwell | g729 |
| 15:08.24 | brunner | does it sound worse than gsm? |
| 15:08.35 | mvanbaak | no |
| 15:08.38 | Qwell | I think it sounds a little better |
| 15:08.42 | Qwell | less bandwidth too |
| 15:08.43 | brunner | hmm |
| 15:08.44 | mvanbaak | g729 sounds better |
| 15:08.51 | brunner | how much bandwidth after overhead? |
| 15:08.59 | Qwell | 20k/s or so I think |
| 15:09.05 | brunner | hmm |
| 15:09.07 | mvanbaak | 19kbps |
| 15:09.11 | mvanbaak | I think |
| 15:09.13 | mvanbaak | lol |
| 15:09.16 | mvanbaak | sorry Qwell |
| 15:09.19 | *** join/#asterisk bkw_ (n=brian@adsl-71-153-162-29.dsl.tul2ok.sbcglobal.net) |
| 15:09.20 | Qwell | it's 8+overhead, and overhead is 12-15ish |
| 15:09.22 | Qwell | I think |
| 15:09.30 | Qwell | bkw_: right? |
| 15:09.38 | tzafrir | brunner, also check ilbc and speex |
| 15:09.41 | mvanbaak | howz everyone |
| 15:09.44 | bkw_ | for? |
| 15:09.49 | Qwell | tzafrir: it'd be unlikely that a provider would support those |
| 15:09.54 | Qwell | bkw_: g729, 20ms |
| 15:10.04 | tzafrir | both are more CPU intensive, but provide nicer quality |
| 15:10.11 | bkw_ | its 20+12 |
| 15:10.19 | Qwell | 20? |
| 15:10.26 | bkw_ | you're talking packet size right? |
| 15:10.28 | tzafrir | and do not reuire licensing overhead |
| 15:10.32 | Qwell | k/s |
| 15:10.38 | bkw_ | yah its 8+ overhead |
| 15:10.45 | Qwell | and overhead is around 12? |
| 15:10.48 | bkw_ | you're better off with 60ms G729 |
| 15:10.48 | *** join/#asterisk qdk (n=qdk@0x573fe57e.bynqu2.broadband.tele.dk) |
| 15:10.55 | bkw_ | 12 is the RTP overhead |
| 15:11.05 | Katty | bkw_: YOU |
| 15:11.15 | Katty | bkw_: why i outta just... |
| 15:11.17 | Katty | bkw_: hug you! |
| 15:11.19 | mvanbaak | does asterisk do 60ms g729 ? |
| 15:11.29 | brunner | so you're telling me that one modern server and a T1 line could support 64 incoming interactive calls? |
| 15:11.42 | Qwell | brunner: pretty much |
| 15:12.02 | brunner | that's incredible. |
| 15:12.05 | tzafrir | brunner, a T1 line supports 23 concurrent calls :-) |
| 15:12.12 | Qwell | tzafrir: data ;) |
| 15:12.18 | tzafrir | ah |
| 15:12.20 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 15:12.27 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 15:12.27 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 15:12.37 | brunner | tzafrir: you mean in voice mode, right? |
| 15:12.52 | brunner | tzafrir: I'm talking about using voip over a T1 |
| 15:14.28 | lmadsen | actros1840: Read() |
| 15:15.48 | [TK]D-Fender | actros1840: You can't |
| 15:16.01 | *** join/#asterisk Lawbringer (n=Lawbring@cpc4-rdng12-0-0-cust292.winn.cable.ntl.com) |
| 15:16.16 | *** join/#asterisk lftsy (n=lftsy@120.194.210.62.te-dns.org) |
| 15:16.19 | dhill | [TK]D-Fender: so if i use Local, it will log both to Master.csv? |
| 15:16.31 | [TK]D-Fender | dhill: Believe so |
| 15:16.47 | *** join/#asterisk bsdwarrior (n=mfahey@fahey.enter.net) |
| 15:16.59 | anonymouz666 | Qwell: I think it's a bit more than that, no? if we consider RTP/UDP/IP/Ethernet overhead |
| 15:17.25 | anonymouz666 | so I would say that a g729 would use around 40kbit/s |
| 15:17.25 | *** join/#asterisk ruied (n=ruied@pal-213-228-184-21.netvisao.pt) |
| 15:20.24 | brunner | there'd be no reason why I couldn't do this from a dedicated server, right? |
| 15:20.28 | JenniferAkemi | i calculated it out at 40 too |
| 15:20.36 | JenniferAkemi | i think there's a chart somewhere |
| 15:20.38 | brunner | run asterisk with voip, I mean |
| 15:21.28 | *** join/#asterisk wmaulik (n=wmaulik@158.59.195.51) |
| 15:22.33 | brunner | never mind, I'm pretty sure that was a stupid question. |
| 15:22.46 | Qwell | brunner: no reason you couldn't |
| 15:22.57 | *** join/#asterisk Havokmon (n=Havokmon@h69-128-237-204.69-128.unk.tds.net) |
| 15:25.13 | outtolunc | his name SErG? |
| 15:25.15 | bsdwarrior | asterisk just crashed out. how can I debug this and find out why? |
| 15:25.42 | mvanbaak | jameswf: that would be.....interesting |
| 15:25.49 | mvanbaak | bsdwarrior: logfiles ? |
| 15:27.16 | actros1840 | [TK]D-Fender : Read() would not help ? |
| 15:27.58 | stansmith | k |
| 15:28.45 | [TK]D-Fender | actros1840: No. |
| 15:29.04 | [TK]D-Fender | actros1840: It will stop on DTMF as well. |
| 15:31.09 | stansmith | b boys will be boys |
| 15:33.51 | actros1840 | [TK]D-Fender : ok, btw : do you know a way to gain a volume while streaming file by MP3Player ? |
| 15:34.16 | [TK]D-Fender | actros1840: Nope |
| 15:34.43 | bsdwarrior | the logs arent show any reason as to why it crashed |
| 15:35.32 | stansmith | bsdwarrior: are you on linux or bsd? |
| 15:36.38 | bsdwarrior | linux |
| 15:37.00 | stansmith | execute dmesg |
| 15:37.21 | stansmith | sift through that and see if you can find anything relative to your asterisk set up |
| 15:38.35 | bsdwarrior | [45521498.920000] atkbd.c: Unknown key released (translated set 2, code 0x6d on isa0060/serio0). |
| 15:38.41 | bsdwarrior | could be nice if linux showed the time |
| 15:39.05 | stansmith | is that referring to an ISA bus? |
| 15:39.50 | bsdwarrior | I have no idea |
| 15:40.05 | bsdwarrior | I dont see anything relevant in dmesg |
| 15:40.17 | *** join/#asterisk angler (n=angler@pdpc/sponsor/digium/angler) |
| 15:40.17 | *** mode/#asterisk [+o angler] by ChanServ |
| 15:40.25 | Katty | hi angler |
| 15:40.43 | stansmith | http://www.google.com/search?&q=atkbd.c |
| 15:40.57 | stansmith | looks like kernel panic, hence why asterisk might of crashed? [confirm/deny] |
| 15:41.25 | Qwell | AT keyboard |
| 15:42.00 | Qwell | no, that's not why asterisk crashed |
| 15:42.58 | anonymouz666 | Katty! |
| 15:43.15 | Katty | ! |
| 15:43.21 | outtolunc | he running xfree86 on that box? |
| 15:43.45 | jameswf | It's one of those rare "perfect" kernels. So if it doesn't happen to compile with your config (or it does compile, but then does unspeakable acts of perversion with your pet dachshund), you can rest easy knowing that it's all your own damn fault, and you should just fix your evil ways. |
| 15:43.45 | jameswf | <PROTECTED> |
| 15:45.21 | stansmith | excuse me? |
| 15:46.30 | jameswf | jbot: excuse stansmith |
| 15:46.33 | *** join/#asterisk mbranca (i=daemon@mi-gw1.voismart.net) |
| 15:46.42 | jameswf | stupid bot |
| 15:46.49 | stansmith | jbot is like 1/5 so far |
| 15:47.12 | *** join/#asterisk KermitTheFragger (n=siepkes@118-197.bbned.dsl.internl.net) |
| 15:48.49 | bsdwarrior | nothing in asterisk log that indicates a crash |
| 15:49.05 | stansmith | bsdwarrior: do you think you might be able to reproduce the crash? |
| 15:49.11 | Katty | jbot: i love you. |
| 15:49.11 | jbot | You love you.? |
| 15:49.16 | Katty | jbot: i love jbot |
| 15:49.17 | jbot | You love jbot? |
| 15:49.20 | Katty | jbot: yes. |
| 15:49.20 | jbot | You don't say! |
| 15:49.22 | *** part/#asterisk dominic1 (n=dob@213.221.82.242) |
| 15:49.27 | jameswf | if kernel pukes before cache is written no logs this is typical you need to set up another kernel as a watchdog |
| 15:50.04 | bsdwarrior | stansmith - it happens maybe every few weeks |
| 15:50.44 | *** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk) |
| 15:52.12 | stansmith | lol |
| 15:52.53 | Katty | i need a course in patience. |
| 15:53.10 | Katty | Patience 101 - Dealing With Stupid People at 8am |
| 15:54.00 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 15:54.00 | *** mode/#asterisk [+o anthm] by ChanServ |
| 15:54.04 | mvanbaak | Katty: that course will last for the rest of your life |
| 15:54.12 | mvanbaak | it's the longest running course they have |
| 15:54.16 | stansmith | graduation is the day you die |
| 15:54.49 | Katty | mvanbaak: that's probably not a bad thing. |
| 15:54.54 | drmessano | or the day someone kills you |
| 15:54.57 | drmessano | Thats an "F" |
| 15:55.05 | stansmith | LOL |
| 15:55.15 | Katty | i think right now i have a C or so |
| 15:55.20 | Katty | people tend to avoid me in the mornings |
| 15:55.48 | drmessano | I imagine I will end up with a D- .. They'll get charged with aggrevated manslaughter and plead out to time served |
| 15:56.24 | drmessano | It will be an episode of Law and Order |
| 15:56.31 | drmessano | "I DID THE WORLD A FAVOR!" |
| 15:57.09 | stansmith | chewebacca defense |
| 15:57.33 | Katty | the support calls aren't what bother me. |
| 15:57.42 | Katty | it's people meeting me in the parking lot to tell me they can't open their email in the morning |
| 15:57.51 | stansmith | LOL that sucks |
| 15:57.52 | Katty | something about that is just... horribly annoying. |
| 15:58.07 | stansmith | not even on the clock, yea |
| 15:58.14 | Katty | oh, it gets worse. |
| 15:58.15 | lnx | stansmith: exten => s-ANSWER,1,AGI(logscript.pl,CALEE_OK,${DIALSTATUS}) ARGV[1] contains value of ${DIALSTATUS} :) |
| 15:58.26 | Katty | I get plenty of phone calls around 9pm from people with questions about their computers at home. |
| 15:58.41 | Katty | Funniest one? Mom called me at work to figure out how to change the input on the tv to VCR |
| 15:58.55 | stansmith | lnx: exten=>s,4,Set(DIGITS=${SWIFT_DTMF}${DIGITS}) |
| 15:59.15 | drmessano | Katty: does anyone VPN from home? |
| 15:59.35 | drmessano | That's the WORST shit ever... Justifiable home tech support because they have VPN |
| 15:59.38 | *** join/#asterisk FlyboySR22 (n=rsears@hq.fw.americanis.net) |
| 15:59.41 | Katty | drmessano: i do. |
| 15:59.51 | Katty | drmessano: but no one knows i do. |
| 15:59.56 | drmessano | The second they install VPN, you're doomed |
| 15:59.56 | Katty | drmessano: except the owner of the company. |
| 16:00.03 | Katty | drmessano: oh, they won't get to. ever. |
| 16:00.08 | drmessano | Oh ok |
| 16:00.12 | drmessano | I have about a dozen that do |
| 16:00.16 | x86 | anyone ever done paging with asterisk? |
| 16:00.16 | Katty | drmessano: the boss has tried to vpn before and knows how much of a hassle it is. |
| 16:00.20 | Katty | drmessano: only techs get it |
| 16:00.24 | Katty | x86: yes, with polycoms |
| 16:00.27 | x86 | not sure if paging is really the correct word... |
| 16:00.32 | nixguy | so does anyone know what software was used for the autoatendat voice in asterisk? |
| 16:00.48 | drmessano | Software? |
| 16:00.52 | Katty | x86: would you like my blog link? |
| 16:00.57 | stansmith | nixguy: I use Cepstral TTS and its pretty good |
| 16:00.57 | nixguy | drmessano: or is it a person? |
| 16:00.58 | x86 | I've got PA speakers in two different zones in my building... I want to play a chime wav file over them at break and lunch times |
| 16:01.05 | drmessano | Its a person |
| 16:01.09 | phillipk | nixguy: A lady named Alison Smith |
| 16:01.16 | nixguy | ok |
| 16:01.19 | nixguy | thnx |
| 16:01.23 | x86 | each zone has different break times, so I'd like to do them independantly |
| 16:01.23 | Katty | x86: ah. |
| 16:01.27 | Katty | x86: nevermind then (= |
| 16:01.30 | lnx | stansmith: i'm talking about perlAGI script get_variable problem. I solved it with ARGV[] ... i have asked u yesterday |
| 16:01.35 | drmessano | http://www.theivrvoice.com/ |
| 16:01.39 | *** join/#asterisk jmacz (n=jmacz@190.84.254.27) |
| 16:01.45 | x86 | Katty: yeah I dont think this is really "paging" per se, more "intercom" heh |
| 16:01.59 | stansmith | lnx: o, i thought we were just showing eachother stuff trying to 1up eachother |
| 16:02.01 | *** join/#asterisk jpeeler (n=jpeeler@216.207.245.1) |
| 16:02.09 | Qwell | Katty: link? |
| 16:02.23 | brunner | are there any voip services that offer unlimited outbound calls to the US for a flat monthly rate? |
| 16:02.23 | x86 | Qwell: ever done something like this before with Asterisk? |
| 16:02.25 | lnx | "And the sign said long haired,freaky people need not apply" |
| 16:02.26 | x86 | not sure it's possible |
| 16:02.28 | Qwell | x86: like what? |
| 16:02.31 | Katty | Qwell: for what? my blog? |
| 16:02.32 | Qwell | and anything is possible |
| 16:02.35 | Katty | Qwell: or that particular post? |
| 16:02.35 | Qwell | Katty: sure |
| 16:02.50 | stansmith | nixguy: http://www.cepstral.com/ |
| 16:02.54 | drmessano | You can link me too... i'm not much of a perv |
| 16:03.03 | Qwell | he's a Dr. |
| 16:03.07 | drmessano | Yes |
| 16:03.12 | stansmith | no hes not |
| 16:03.12 | x86 | Qwell: intercom... basically I've got 2 PA speakers in separate zones in the building... at certain times of the day (different for both zones), I want to play a chime sound over the speaker in the specified zone |
| 16:03.15 | drmessano | ~drmessano |
| 16:03.16 | jbot | [drmessano] the leading cause of censorship in #asterisk, maybe a Doctor, not really a mess um a no..... The Doctor is accepting new patients, <drmessano> I am not a OB/GYN but I'll have a look anyway |
| 16:03.23 | drmessano | I'll have a look anyway |
| 16:03.30 | stansmith | lol |
| 16:03.36 | Qwell | x86: call file |
| 16:03.46 | x86 | file: you around? |
| 16:03.55 | x86 | Qwell: got his digits? :) |
| 16:04.01 | file | O.o |
| 16:04.03 | Qwell | .call file |
| 16:04.10 | Qwell | ... |
| 16:04.16 | drmessano | Yeah, lets give out developers home phone numbers on IRC |
| 16:04.19 | brunner | Qwell: are there any voip services that offer unlimited outbound calls to the US for a flat monthly rate? |
| 16:04.22 | x86 | Qwell: how do i interface the speakers with Asterisk? |
| 16:04.23 | drmessano | THAT sounds like a good idea |
| 16:04.30 | Qwell | x86: chan_alsa, chan_oss, chan_console |
| 16:04.36 | Qwell | app_jack |
| 16:04.38 | x86 | drmessano: was mainly asking for a SIP/IAX number... and probably at work, not home ;) |
| 16:04.46 | Katty | Qwell: http://angela.sleekgeek.org/2007/11/06/freepbx-paging-and-intercom/ |
| 16:04.48 | nixguy | stansmith: thnx! hehe im even tempted to rip it right of their homepage :) |
| 16:04.49 | x86 | Qwell: what hardware do I need? :) |
| 16:05.02 | Qwell | I don't know what kind of speakers they are |
| 16:05.11 | x86 | Qwell: two-wire speakers |
| 16:05.26 | Qwell | get them into 3.5mm, and use a soundcard |
| 16:05.29 | stansmith | nixguy: theres actually a well known way to circumvent their licensing, i brought it to their attention and they told me they are aware of it and working on fixing it |
| 16:05.32 | drmessano | Oh yes.. A SIP/IAX number is nothing at all like a PSTN number.... sure, give it out |
| 16:05.37 | Katty | Qwell: it's not precisely for straight up asterisk, but close enough. |
| 16:05.49 | x86 | Qwell: would I need two sound cards (one for each zone) ? |
| 16:06.01 | Qwell | x86: probably |
| 16:06.01 | bsdwarrior | which .so should I reload to reload the echo cancellers ? |
| 16:06.20 | x86 | Qwell: Qwell would you recommend any of those channel drivers over the others? |
| 16:06.21 | bsdwarrior | instead of a "reload". when I do a "reload", phones starting ringing with two calls |
| 16:06.28 | *** join/#asterisk gutzb4ll (n=mark@152.160.35.1) |
| 16:06.36 | gutzb4ll | hello |
| 16:06.38 | Qwell | x86: whatever your system can already do.. |
| 16:06.46 | Qwell | it doesn't really matter for simple things like that |
| 16:06.51 | gutzb4ll | is there any way for me to create a new field that will be populated in the CDR database? |
| 16:06.52 | nixguy | stansmith: hehe tempting |
| 16:06.58 | *** join/#asterisk ThoMe (n=tm@81.92.168.130) |
| 16:07.00 | ThoMe | hello |
| 16:07.02 | ThoMe | :-) |
| 16:07.04 | gutzb4ll | i would like a field containing the uniqueid of the call as seen from within the dialplan |
| 16:07.31 | x86 | Qwell: hmm... never really used any of those, except to make outbound calls with chan_oss from the CLI for testing |
| 16:08.02 | *** join/#asterisk lesouvage (n=lesouvag@92.65.174.153) |
| 16:08.21 | stansmith | ~hello ThoMe |
| 16:08.22 | jbot | ACTION stands up and shouts "HELLO ThoMe!" |
| 16:08.51 | ThoMe | :-) |
| 16:09.03 | stansmith | jbot thanks |
| 16:09.03 | jbot | bitte, stansmith |
| 16:09.32 | x86 | Qwell: is there a howto on basically what I want to do? |
| 16:09.38 | ThoMe | Wenn ein Anruf eingeht landet er erstmal in der Queue. Ein Agent nimmer ab und merkt "ist nicht für mich, ich verbinde weiter", kann ich in der CDR-Datenbank sehen wohin er verbunden hat? |
| 16:09.41 | x86 | Qwell: specifically, how do i use the channel driver appropriately |
| 16:09.47 | Qwell | just dial it |
| 16:09.54 | x86 | Qwell: Dial(OSS/foo)? |
| 16:09.59 | Qwell | pretty much |
| 16:10.41 | Qwell | see oss.conf |
| 16:11.21 | lnx | ~wardial |
| 16:12.22 | brunner | Qwell: how can I figure out how many calls one server can handle? |
| 16:12.30 | Qwell | brunner: try it |
| 16:12.40 | brunner | seems like it'd be kind of hard to test |
| 16:13.04 | x86 | Qwell: think this may help: http://www.voip-info.org/wiki/view/Setting+up+paging+with+a+sound+card |
| 16:15.18 | x86 | Qwell: I'm confused why oss.conf needs a context |
| 16:15.32 | x86 | Qwell: is that if I'm using it for inbound calls? |
| 16:16.02 | ThoMe | incomming calls are put in the queue first. An agent always picks up and realizes "not for me, im transfering". Can i see in the CDR database where he |
| 16:16.05 | ThoMe | refered the call to? |
| 16:16.28 | *** join/#asterisk l2cache (n=l2cache@64.128.254.98) |
| 16:19.29 | *** join/#asterisk VaNNi (n=VaNNi@216.70.165.200) |
| 16:20.57 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 16:20.57 | *** mode/#asterisk [+o russellb] by ChanServ |
| 16:21.15 | drmessano | Ah, so that's russellb's home number... thanks for posting it :) |
| 16:21.40 | gutzb4ll | is anyone familiar with the CDR backend for Asterisk? how much coding would be required to add another field to the CDR log? |
| 16:22.28 | russellb | drmessano: i know you are lying! |
| 16:22.34 | russellb | i don't have a home phone. |
| 16:22.52 | Qwell | surely you didn't think asterisk developers would actually use phones |
| 16:22.55 | x86 | Qwell: all of these examples I'm finding say Dial(console/dsp), Playback, Hangup |
| 16:23.02 | russellb | yeah, i hate phones |
| 16:23.02 | Qwell | x86: see the config |
| 16:23.05 | gutzb4ll | i see in the code for the CDR plugin that there is an ifdef for logging the unique id |
| 16:23.08 | x86 | Qwell: how do i say dsp0 or dsp1? |
| 16:23.10 | gutzb4ll | but i dont see how it would be set |
| 16:23.11 | Qwell | x86: see the config |
| 16:23.13 | x86 | Qwell: which config? |
| 16:23.16 | Qwell | oss.conf |
| 16:23.21 | drmessano | lol |
| 16:23.21 | x86 | oss.conf doesn't specify it |
| 16:23.25 | Qwell | yes it does |
| 16:23.28 | x86 | please look |
| 16:23.32 | Qwell | I just did |
| 16:23.43 | drmessano | I've always said the Asterisk devs know nothing about phones... now, I have my proof |
| 16:23.47 | x86 | pastebin yours then, it must be non-standard |
| 16:23.54 | stansmith | gutzb4ll: are you trying to get ${UNIQUEID} ? |
| 16:23.59 | Qwell | I'm staring at the sample config.. |
| 16:24.03 | x86 | me too |
| 16:24.11 | x86 | says nothing about multiple sound cards |
| 16:24.11 | Qwell | upgrade to 1.4 |
| 16:24.25 | x86 | uh, I am? |
| 16:24.25 | *** join/#asterisk supjigator (n=shanebur@152.53.16.10) |
| 16:24.34 | stansmith | gutzb4ll: i just bypass all that and pass any variables i want to log as CDR to php which in turn places it in postgres |
| 16:24.35 | x86 | 1.4.12.1 ;) |
| 16:24.48 | Qwell | look at the sample config shipped with it |
| 16:24.56 | x86 | dude, I am! |
| 16:25.01 | Qwell | configs/oss.conf |
| 16:25.02 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 16:25.11 | gutzb4ll | stansmith: yes, i want it to be in the CDR database. You see, im trying to make a web frontend for listening to recorded calls. the calls are referenced by the ${uniqueid} of the call they came from |
| 16:25.14 | x86 | hold on, I'll patebin the sample that I have |
| 16:25.23 | seanbright | Qwell/x86: http://svn.digium.com/view/asterisk/tags/1.4.12.1/configs/oss.conf.sample?revision=84779&view=markup |
| 16:25.43 | Qwell | ^^ shows multiple cards |
| 16:25.55 | *** join/#asterisk Yourname`` (n=chatzill@unaffiliated/yourname/x-837320) |
| 16:26.09 | stansmith | gutzb4ll: yea so just pass ${UNIQUEID} and anything else to a php (or what have you) and do whatever with it |
| 16:26.11 | Yourname`` | So, really. Does anyone know why my email to the list is not showing up? |
| 16:26.22 | x86 | Qwell: http://pastebin.ca/929181 <-- sample that comes with 1.4.12.1 |
| 16:26.24 | stansmith | to my knoweldge, i dont think you can change ${UNIQUEID}, only read it, though i could be wrong |
| 16:26.28 | Qwell | x86: wrong :) |
| 16:26.41 | Qwell | that's from 1.2 |
| 16:26.48 | x86 | hmm |
| 16:26.57 | Qwell | configs/oss.conf |
| 16:27.02 | seanbright | x86: this is the config from 1.4.12.1 --> http://svn.digium.com/view/asterisk/tags/1.4.12.1/configs/oss.conf.sample?revision=84779&view=markup |
| 16:27.07 | x86 | perhaps when I upgraded from 1.2.21 some time ago, it didn't upgrade the samples ;) |
| 16:27.10 | seanbright | from the horses mouth, so to speak. |
| 16:27.13 | drmessano | I could read the URL and see that... doh |
| 16:28.21 | x86 | Qwell: what do I set the "device" to? |
| 16:28.36 | x86 | Qwell: I see multiple card definitions at the end, but it wants an I/O device |
| 16:29.19 | *** join/#asterisk jmesquita (n=jmesquit@201.7.117.114) |
| 16:30.26 | *** join/#asterisk CunningPike (n=arodgers@204.239.12.183) |
| 16:30.28 | x86 | Qwell: also, how do I dial the different cards? |
| 16:32.09 | [TK]D-Fender | x86: s/it/I/ |
| 16:32.34 | *** join/#asterisk joshaidan (n=brianj@S0106001c1023e838.tb.shawcable.net) |
| 16:32.38 | x86 | [TK]D-Fender: ? |
| 16:33.20 | *** join/#asterisk iCEBrkr (n=icebrkr@cyberdyne.org) |
| 16:33.43 | x86 | all of the examples google can find me only use one sound card |
| 16:33.45 | [TK]D-Fender | x86: For your "samples" issue |
| 16:34.07 | *** join/#asterisk iCEBrkr (n=icebrkr@cyberdyne.org) |
| 16:34.46 | ThoMe | Qwell: incomming calls are put in the queue first. An agent always picks up and realizes "not for me, im transfering". Can i see in the CDR database where he refered the call to? |
| 16:34.56 | x86 | Qwell: found far less information regarding chan_alsa, and almost nothing for chan_jack |
| 16:35.20 | ThoMe | Qwell: i can see where he refered the call to.. but not where have recieve the call |
| 16:35.32 | ThoMe | Qwell: do you have ideas? |
| 16:35.56 | stansmith | hillary clinton is a fox for real |
| 16:36.40 | x86 | Qwell: and chan_console isn't even available for 1.4 |
| 16:38.19 | russellb | there is no such thing as chan_jack :) |
| 16:38.36 | ThoMe | russellb: hello. do you have a idea / answer to my question? |
| 16:38.52 | russellb | i have absolutely no ideas about anything |
| 16:39.07 | ThoMe | russellb: :-)) |
| 16:39.18 | x86 | russellb: google says there is |
| 16:39.52 | russellb | russellb: app_jack |
| 16:39.57 | x86 | ah |
| 16:39.59 | russellb | why am i talking to myself? |
| 16:40.05 | stansmith | lol |
| 16:40.05 | russellb | but yeah, app_jack ... |
| 16:40.06 | x86 | good question :) |
| 16:40.26 | x86 | russellb: will that do multiple zone paging with PA speakers hooked up to two different sound cards? |
| 16:40.28 | russellb | http://www.russellbryant.net/blog/index.php/2008/01/13/jack-interfaces-for-asterisk/ |
| 16:40.31 | Yourname`` | Mawnin' russellb |
| 16:40.42 | russellb | x86: yeah, that would be easy to do with jack |
| 16:40.49 | russellb | except it's only in asterisk 1.6 |
| 16:40.53 | russellb | Yourname``: greetings |
| 16:40.57 | Katty | hi russellb! |
| 16:40.57 | x86 | gah |
| 16:41.15 | russellb | x86: however, here's a hack for you, regardless of what version you're using ... |
| 16:41.23 | russellb | look into the jackoss interface |
| 16:41.35 | russellb | when lets you use OSS as an interface into jack |
| 16:41.45 | russellb | and then you could configure your jack ports to go to 2 sound cards |
| 16:41.46 | seanbright | jackoss? for serious? |
| 16:41.50 | russellb | for serious |
| 16:41.55 | seanbright | the double entendres abound |
| 16:41.56 | russellb | ossjack |
| 16:41.59 | russellb | i don't know what it's called |
| 16:42.20 | *** join/#asterisk ZeRoCoOl_ (n=ZeRoCoOl@41.225.140.234) |
| 16:42.29 | ZeRoCoOl_ | hi |
| 16:42.32 | russellb | oss2jack |
| 16:42.43 | russellb | anyway, yeah, do that. that would do it. :) |
| 16:43.14 | ZeRoCoOl_ | please I want to to change my recording folder |
| 16:43.22 | russellb | ok |
| 16:43.28 | seanbright | we'll allow it |
| 16:43.29 | seanbright | heh |
| 16:43.32 | Qwell | I won't |
| 16:43.35 | russellb | ;p; |
| 16:43.37 | russellb | lol, too |
| 16:43.42 | Qwell | ;p; works |
| 16:43.46 | ZeRoCoOl_ | from /var/spool/asterisk/monitor to another location |
| 16:43.46 | ZeRoCoOl_ | lol |
| 16:43.54 | russellb | ok. |
| 16:43.55 | x86 | russellb: no way to do this with chan_oss like Qwell suggested? |
| 16:43.57 | Qwell | ZeRoCoOl_: so give Record() a path |
| 16:44.11 | russellb | x86: not in asterisk 1.2 ... |
| 16:44.17 | x86 | russellb: 1.4? |
| 16:44.23 | russellb | even in 1.4 where it supports multiple oss devices, it would be tough, i think |
| 16:44.29 | x86 | Qwell: 10:41 <@Corydon76-dig> chan_oss only supports a single soundcard |
| 16:44.30 | seanbright | ZeRoCoOl_: or change astspooldir in asterisk.conf |
| 16:44.39 | russellb | x86: actually, yes, in 1.4, it would work |
| 16:44.44 | x86 | russellb: oh? |
| 16:44.49 | Qwell | that's what I said! |
| 16:44.49 | x86 | russellb: got a tutorial? :) |
| 16:44.52 | russellb | x86: using the Page() application with multiple OSS channels |
| 16:44.55 | ZeRoCoOl_ | I don't know where I can find the fonction... I looked everywhere no way... some calls a registred automatically in "monitor folder"... |
| 16:44.56 | russellb | x86: pfft ... no :-p |
| 16:45.00 | x86 | hahah |
| 16:45.07 | x86 | ok, I'll check out Page() |
| 16:45.09 | x86 | thanks :) |
| 16:45.12 | russellb | oh |
| 16:45.15 | seanbright | ZeRoCoOl_: read /etc/asterisk/asterisk.conf |
| 16:45.20 | seanbright | ZeRoCoOl_: and look for astspooldir |
| 16:45.22 | russellb | and set all of your oss channels in oss.conf to autoanswer |
| 16:45.27 | ZeRoCoOl_ | I found /var/spool/asterisk |
| 16:45.36 | seanbright | ZeRoCoOl_: read /etc/asterisk/asterisk.conf |
| 16:45.38 | x86 | russellb: did that already |
| 16:45.38 | seanbright | ZeRoCoOl_: and look for astspooldir |
| 16:45.43 | russellb | ZeRoCoOl_: specify an absolute path to Monitor() / MixMonitor() |
| 16:45.48 | russellb | x86: cool, then yeah, Page() is your friend |
| 16:46.18 | x86 | russellb: so what do I pass for the channel to Page? |
| 16:46.25 | russellb | app_jack would be cooler, though .. but only because i wrote it :-p |
| 16:46.31 | x86 | russellb: how do i specify dsp0 and dsp1? |
| 16:46.38 | russellb | x86: OSS/device1 and OSS/device2 |
| 16:46.46 | russellb | and you configure those devices in oss.conf |
| 16:46.48 | ZeRoCoOl_ | so /var/spool/asterisk is the absolute folder for Monitor() ? so why he record on "monitor" folder |
| 16:46.49 | russellb | configs/oss.conf.sample |
| 16:47.00 | x86 | russellb: yep, qwell got me that far :) |
| 16:47.03 | russellb | ZeRoCoOl_: it isn't, it's the base dir for a lot of things |
| 16:47.07 | x86 | cool deal |
| 16:47.30 | russellb | hot deal |
| 16:47.33 | x86 | thanks a bunch :) |
| 16:47.37 | x86 | ok Paris |
| 16:47.38 | Qwell | luke-warm deal |
| 16:47.41 | russellb | surely |
| 16:47.43 | russellb | now that will be $20 |
| 16:47.48 | x86 | bahahaha |
| 16:48.04 | x86 | how about $2.50 and a jawbreaker? |
| 16:48.10 | Qwell | dibs on the jawbreaker |
| 16:48.11 | russellb | o.O |
| 16:48.13 | russellb | lol |
| 16:48.18 | Qwell | ^^ true |
| 16:48.21 | stansmith | ew |
| 16:48.25 | russellb | surely, i'll take it |
| 16:48.44 | russellb | x86: our mailing address is on digium.com ... we'll be looking for the package |
| 16:48.48 | ZeRoCoOl_ | russellb if isn't what is the name of the monitoer folder variable ? |
| 16:48.50 | robmac67 | does anyone know how to specify the stun server in gtalk/jabber.conf ? |
| 16:48.51 | ZeRoCoOl_ | astspooldir => /var/spool/asterisk <= ? |
| 16:48.52 | russellb | x86: i would like my $2.50 in nickels, please |
| 16:49.12 | x86 | russellb: rofl... |
| 16:49.15 | Qwell | I wonder what would happen if somebody shipped something to Digium, c/o: Qwell |
| 16:49.21 | russellb | ZeRoCoOl_: specify an _absolute_ path to Monitor / MixMonitor and you're done |
| 16:49.31 | Qwell | how long do you think it would take to reach me? I'm betting fairly quickly. |
| 16:49.34 | russellb | Qwell: Qwell c/o Digium |
| 16:49.41 | Qwell | umm, that too |
| 16:49.50 | Qwell | you know what I meant though :D |
| 16:49.54 | russellb | yeah :) |
| 16:49.57 | russellb | just trollin' |
| 16:51.41 | ZeRoCoOl_ | russellb I have to do it in asterisk.conf ? |
| 16:51.54 | russellb | nooooooo |
| 16:51.59 | russellb | ZeRoCoOl_: did you write your dialplan? |
| 16:52.01 | ZeRoCoOl_ | in my dialplan ? |
| 16:52.05 | russellb | yes |
| 16:52.25 | seanbright | heads |
| 16:52.29 | russellb | tails |
| 16:52.30 | Qwell | ~freepbx |
| 16:52.31 | jbot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 16:52.38 | seanbright | i win! |
| 16:52.45 | seanbright | what do i win? another beeeer |
| 16:52.47 | *** join/#asterisk Bonix-BR (n=Bonix-BR@217-lo1.rt2.isimples.com.br) |
| 16:52.48 | x86 | russellb: so after I Page(), can I Playback? |
| 16:52.50 | russellb | yay beer |
| 16:52.55 | russellb | x86: no |
| 16:52.56 | ZeRoCoOl_ | I don't have the Monitor/Mixmonitor function in my dialplan... I think that FeePBX is doing this... |
| 16:53.01 | Qwell | I WIN |
| 16:53.03 | Qwell | WHAT NOW |
| 16:53.04 | x86 | russellb: then it's not a solution? :( |
| 16:53.06 | russellb | Qwell++ |
| 16:53.06 | seanbright | Qwell wins!!! |
| 16:53.07 | russellb | Qwell++ |
| 16:53.15 | russellb | x86: you can, just not like that |
| 16:53.26 | seanbright | jbot: Qwell++ |
| 16:53.28 | russellb | x86: you have to originate a call between a playback and the page app |
| 16:53.35 | x86 | well sure |
| 16:53.37 | x86 | using a call file |
| 16:53.42 | x86 | no problem |
| 16:53.43 | russellb | yeah |
| 16:53.49 | russellb | so if you do that, then you're good |
| 16:54.08 | x86 | my question is that once the Page() is executed, can I Playback() and have it go to that channel? |
| 16:54.51 | *** join/#asterisk juanjoc (n=juanjoc@200.69.219.113) |
| 16:54.53 | *** join/#asterisk susinths (n=susinths@sos3-1x-dhcp127.studby.uio.no) |
| 16:55.08 | ZeRoCoOl_ | I looked in some agi scripts in my *.conf files noway the only Macro whois doing that, is "macro-record-enable" I changed the twoo values but noway... |
| 16:55.28 | seanbright | ~freepbx |
| 16:55.29 | jbot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 16:55.34 | seanbright | ZeRoCoOl_: read that ^^^ |
| 16:55.55 | ZeRoCoOl_ | ah ok |
| 16:56.02 | ZeRoCoOl_ | thank you |
| 16:56.11 | ZeRoCoOl_ | russellb thinks a lot |
| 16:56.16 | seanbright | he does |
| 16:56.18 | ZeRoCoOl_ | thanks* |
| 16:56.21 | seanbright | we hope |
| 16:59.16 | robmac67 | does anyone know how to specify a stun server in gtalk/jabber.conf ? |
| 16:59.26 | ThoMe | Qwell: can you help me? :-) |
| 16:59.31 | ThoMe | you |
| 17:00.09 | x86 | what's that sox command for converting a wav file with X sample rate down to mono 8000Hz sample rate suitable for playing with asterisk? |
| 17:00.29 | seanbright | x86: http://www.voip-info.org/tiki-index.php?page=Convert+WAV+audio+files+for+use+in+Asterisk |
| 17:01.38 | x86 | sox: Failed reading cuckoo.wav: Sorry, this WAV file is in MPEG Layer 3 format. |
| 17:01.41 | x86 | grr |
| 17:02.04 | *** join/#asterisk ruied (n=ruied@pal-213-228-184-21.netvisao.pt) |
| 17:02.54 | juanjoc | file: Are you available? I've run some more tests with the intended fix for ticket #11491. I can confirm that the line saying "Packet2Packet bridging..." does not appear when the call fails. |
| 17:02.56 | stansmith | steve jobs is a crazy ass hole |
| 17:03.30 | x86 | can someone convert this to sln format for me: http://www.wavsource.com/snds_2008-03-01_888350906694402/sfx/cuckoo_clock2_x.wav |
| 17:03.31 | seanbright | a _rich_ crazy asshole |
| 17:03.38 | russellb | stansmith: are you a bot? |
| 17:03.46 | Yourname`` | All crazy assholes are rich. |
| 17:03.48 | stansmith | russellb: 1 |
| 17:03.50 | Qwell | x86: convert cuckoo_clock2_x.wav cuckoo_clock2_x.sln |
| 17:04.00 | *** mode/#asterisk [+b %stansmith!*@*] by russellb |
| 17:04.34 | seanbright | heh... seriously? if (msg == "are you a bot?") { return 1; } |
| 17:04.48 | Yourname`` | LOL |
| 17:04.59 | x86 | Unable to open input file: /var/lib/asterisk/sounds/cuckoo.wav |
| 17:05.12 | x86 | Qwell: sox doesn't like it and asterisk doesn't like it |
| 17:05.21 | seanbright | because its an MP3 with a .wav extension |
| 17:05.30 | seanbright | or did i miss something? |
| 17:05.35 | x86 | seanbright: no, it's not |
| 17:05.42 | x86 | cuckoo.wav: RIFF (little-endian) data, WAVE audio, MPEG Layer 3, mono 11025 Hz |
| 17:05.55 | x86 | wave audio |
| 17:06.07 | seanbright | ohhhhh |
| 17:06.09 | x86 | not sure what the mpeg layer 3 part is for |
| 17:06.14 | seanbright | i thought "MPEG Layer 3" meant "mp3" |
| 17:06.15 | x86 | mpg123 can't play it |
| 17:06.39 | x86 | seanbright: it's using mp3 for something, but the actual audio is wave |
| 17:06.58 | x86 | perhaps the codec is mp3 and the container is wav |
| 17:07.07 | x86 | Qwell: can you convert it for me? |
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| 17:07.26 | x86 | pa pa pa pwease? :) |
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| 17:08.30 | *** mode/#asterisk [+o angler] by irc.freenode.net |
| 17:08.41 | *** join/#asterisk BCS-Satori (n=somewher@75.148.21.114) |
| 17:09.48 | vap0rtranz | anyone recommend a graphical call accounting frontend? .. for business types the enjoy actually charging for calls |
| 17:10.23 | *** join/#asterisk plik (i=gorph@phalse.2600.COM) |
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| 17:12.51 | jamesrdorn | ugh, not again |
| 17:12.51 | outtolunc | ride'm cowpoke |
| 17:12.51 | jamesrdorn | another split due to DDOS? |
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| 17:12.53 | vap0rtranz | ummm ... voip-info.org/wiki having problems? |
| 17:12.58 | x86 | vap0rtranz: a2billing? |
| 17:13.38 | ZeRoCoOl_ | vap0rtranz yeah |
| 17:13.45 | ZeRoCoOl_ | oh no |
| 17:13.47 | ZeRoCoOl_ | all right now |
| 17:13.48 | Yourname`` | Anyone think using Asterisk to dial via callfiles on a Dell 1850 is overkill? |
| 17:13.50 | jamesrdorn | vap0rtranz, slow, but works |
| 17:13.53 | Juggie | vap0rtranz, http://mirror.su.lt/voip-info/wiki/view/Voip-Info+Mirrors.html |
| 17:14.42 | vap0rtranz | jamesrdorn: don't ya hate that. it's back |
| 17:14.47 | vap0rtranz | Juggie: ty |
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| 17:14.59 | jamesrdorn | vap0rtranz, =) |
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| 17:16.11 | vap0rtranz | x86: they even have a demo! awesome, ty |
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| 17:18.43 | jamesrdorn | =( |
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| 17:19.18 | *** mode/#asterisk [+o anthm] by ChanServ |
| 17:19.26 | *** join/#asterisk esaym (n=user@cpe-72-183-198-134.satx.res.rr.com) |
| 17:19.51 | BCS-Satori | I have an issue with asterisk 1.4.11 through 1.4.18. I need to know why when I loose communication with an ISTP trunk, the whole system significate delays, 10-20 seconds, on phone calls Internal & Externally. For example, we tested this by removing our Internet Connection, not the router, from our network and reloaded asterisk to find out that we can no longer make internal phones without a 10-20 second delay. |
| 17:19.52 | BCS-Satori | If the user attempts to make an internal call the call never connects and when the user hangs up the phone, the reciepeint of the call begins to ring after 10-20 seconds with the CID as the calling phone which of course is no longer calling, causing a ghost phone call. When this occurs we reiceve a message stating: "chan_sip.c:1947 in retrans_pkt:.." |
| 17:20.56 | BCS-Satori | This is pretty scary considering the the internet conenction fails this occurs and the whole system begins unusable |
| 17:21.04 | *** join/#asterisk Washy (i=Washy@gateway/tor/x-8465e4deaa2d9208) |
| 17:21.54 | Washy | Does anyone know of a number you can call to hear your DID number read back to you |
| 17:22.10 | supjigator | 8285871811 |
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| 17:22.17 | *** mode/#asterisk [+ooo putnopvut Qwell twisted] by irc.freenode.net |
| 17:22.47 | jamesrdorn | neat |
| 17:22.59 | *** part/#asterisk jivco (n=jivco@85.187.217.6) |
| 17:23.08 | Washy | Better not be a porn line |
| 17:23.08 | supjigator | whats that? |
| 17:23.09 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 17:23.14 | jamesrdorn | anyone notice that trixbox pro iso will not start in VMware Workstation? |
| 17:23.22 | jamesrdorn | Washy, it works |
| 17:24.02 | gutzb4ll | does anyone know how to solve this error: format_mp3.c:46: error: redefinition of ‘struct ast_filestream’ |
| 17:24.02 | gutzb4ll | im trying to compile asterisk_addons |
| 17:24.30 | supjigator | Washy: Its not |
| 17:24.31 | supjigator | Its a ANI line |
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| 17:25.29 | *** mode/#asterisk [+o Qwell] by ChanServ |
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| 17:26.00 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 17:27.29 | vap0rtranz | gutzb4ll: sounds like some library mismatch |
| 17:27.42 | gutzb4ll | yea, thats what im afraid of. |
| 17:27.44 | *** join/#asterisk adjohn (n=adjohn@p5155-ipad74marunouchi.tokyo.ocn.ne.jp) |
| 17:29.08 | vap0rtranz | gutzb4ll: i just gave up on playing mp3. all wav/gsm here |
| 17:29.59 | robmac67 | BCS-Satori: have you got srvlookup=yes in sip.conf ? |
| 17:30.53 | *** join/#asterisk jetlagmk2 (i=jetlag@70.17.48.245) |
| 17:31.59 | BCS-Satori | robmac67: turned off, we ran into that issue a long time ago |
| 17:34.53 | twisted | hmm |
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| 17:35.20 | Washy | you know how pay phones have the wrong number on them most of the time? |
| 17:35.54 | *** join/#asterisk Schreiber1337 (n=chatzill@static-71-116-30-94.eriepa.dsl-w.verizon.net) |
| 17:36.07 | *** part/#asterisk ZeRoCoOl_ (n=ZeRoCoOl@41.225.140.234) |
| 17:36.13 | Washy | why is that? |
| 17:36.26 | Washy | so that you can't have someone call you and avoid usurious rates? |
| 17:38.06 | rickross | anyone ever experience weird dropouts in a MeetMe conference? (sounds like the system is squelching what it thinks is silence) |
| 17:38.13 | *** part/#asterisk brunner (n=chris@66.35.172.123) |
| 17:38.19 | jamesrdorn | Washy, most payphones do not accept incoming calls anymore |
| 17:38.34 | rickross | it seems like the MeetMe app is suppressing audio at the beginning and end of when people are speaking |
| 17:38.35 | jamesrdorn | Washy, and... I havent seen a payphone in a long while |
| 17:39.05 | rickross | this is in 1.6b4 - dunno if that is a new behavior, or if we're seeing a bug of some sort? |
| 17:39.21 | Washy | I've seen them get incoming calls quite recently |
| 17:39.41 | Washy | jamesrdorn: Most gas stations still have them |
| 17:39.49 | Schreiber1337 | Has anyone ever created a way to have someone call in, record a message and then have the Asterisk box call a long list of phone numbers and play the message back when they pick up? Kind of a Emergency Broadcast system…. |
| 17:39.57 | Washy | THing is, it's always a wrong number |
| 17:40.22 | Washy | So there may be *some* number that maps to them |
| 17:40.27 | juanjoc | file: ping |
| 17:41.04 | jamesrdorn | Washy, the truth is, if telcos were smart they would not send the CALLERID info from payphones |
| 17:42.05 | Washy | But you don't know for a fact that incoming calls are impossible, right? |
| 17:42.46 | jamesrdorn | Washy, well incoming calls are possible for some of them that I do know. But I would think most pay phones would have that turned off. |
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| 17:46.31 | *** join/#asterisk flujan (n=flujan@200-160-115-020.static.spo.ctbc.com.br) |
| 17:46.50 | Washy | I miss payphones |
| 17:46.55 | vap0rtranz | a2billing is actually too big. i should have said a gui that the local office manager can check her call records; not for voip resellers |
| 17:47.21 | *** join/#asterisk susinths_ (n=susinths@sos3-1x-dhcp127.studby.uio.no) |
| 17:47.33 | jpeeler | Washy: have you ever tried to call 18004444444? |
| 17:47.47 | jpeeler | that's MCI and they read you back the number your calling from |
| 17:47.51 | Washy | no |
| 17:47.53 | *** join/#asterisk BipBip (n=BipBip@194.65.5.235) |
| 17:47.54 | jpeeler | *you're |
| 17:47.58 | Washy | should I |
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| 17:48.26 | jpeeler | well you said that the number on the payphone wasn't correct, i was curious if it would match what was on the payphone |
| 17:49.15 | Washy | I'll keep that in mind |
| 17:49.23 | *** join/#asterisk atis_work (n=atis_wor@81.198.164.2) |
| 17:49.30 | jpeeler | certainly an easy number to remember |
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| 17:52.27 | cmantito | hahaha |
| 17:52.29 | cmantito | errr oops |
| 17:52.43 | cmantito | so, I left my desk this morning, I told ast to forward my calls to my cellphone |
| 17:52.57 | cmantito | I got home, I forgot to unforward my calls, but as my routine usually is, I told my cellphone to forward my calls to my DID |
| 17:53.09 | cmantito | yay loops :D |
| 17:53.44 | cmantito | when I finally unforwarded, my sip phone started ringing with 12 incoming calls that had sat there looping that I had to hangup |
| 17:54.10 | cmantito | remember kids: only YOU can prevent calling loops |
| 17:56.53 | Strom_C | yeah, by writing error-checking routines into your system that return a congestion or busy condition if such a thing occurs :P |
| 17:57.39 | *** join/#asterisk flujan (n=flujan@200-160-115-020.static.spo.ctbc.com.br) |
| 17:57.44 | flujan | #asterisk-dev |
| 17:58.05 | cmantito | i don't see it as a particularly easy thing to track |
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| 18:00.31 | jameswf | ~ |
| 18:01.58 | *** join/#asterisk tecnico (n=tecnico@user-24-214-56-217.knology.net) |
| 18:04.21 | *** join/#asterisk DonAlex (n=DonAlex@mirafiori.demon.co.uk) |
| 18:04.26 | DonAlex | Evening all. |
| 18:04.44 | *** join/#asterisk angryuser (n=nononon@df01t2-213-44-80-96.d4.club-internet.fr) |
| 18:04.46 | Washy | So ordinarily what you do is order a DID number from a service provider, the configure your asterisk server to route the call to a local client |
| 18:05.09 | Washy | and you can configure the astirsk server about the properties of the call? |
| 18:05.13 | DonAlex | Anyone have any idea what "span1 got hang up request , cause 90 " means? |
| 18:05.59 | [TK]D-Fender | Washy: huh?! |
| 18:06.10 | *** join/#asterisk nvrpunk (n=root@81.90.21.227) |
| 18:06.29 | supjigator | DonAlex: Lookup cause 90. Likely cause code depending what span 1 is. |
| 18:08.36 | Katty | i are teh stuffed. |
| 18:08.52 | DonAlex | supjigator: ahh hang on I can tell you.. |
| 18:08.55 | Washy | I'm just trying to understand why a normal user would use an asterisk server |
| 18:09.14 | Katty | because it coudl be fun? |
| 18:09.34 | supjigator | Cause for $1000 or less you can replace a $25,000 PBX |
| 18:09.36 | *** join/#asterisk lvl- (n=lvl@101.248.dhcp.lht.hhs.nl) |
| 18:09.38 | *** join/#asterisk CVirus (n=GoD@81.10.29.145) |
| 18:09.40 | Washy | What things can I do with it? |
| 18:09.42 | DonAlex | supjigator: PRI ISDN line. |
| 18:09.48 | jameswf | because ron paul says so |
| 18:09.52 | supjigator | google PRI cause code. |
| 18:10.02 | Katty | Washy: feed cats. |
| 18:10.03 | supjigator | Cisco has em somewhere. |
| 18:10.06 | Washy | who is ron paul? |
| 18:10.08 | jameswf | ~book |
| 18:10.09 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 18:10.10 | DonAlex | supjigator: Oh it is a PRI code? |
| 18:10.13 | jameswf | Washy: ^^^^^^^^^^^^^^^^^^^6 |
| 18:10.22 | Katty | Washy: Ron Paul is running for president. |
| 18:10.26 | Katty | Washy: he won't get it tho. |
| 18:10.28 | supjigator | PRI cause code |
| 18:10.40 | Katty | Washy: As far as I know, he didn't get any states. |
| 18:10.50 | jameswf | ron paul can use chuck norris tears to cure cancer |
| 18:10.59 | Washy | I think that's Ralph Nader |
| 18:11.16 | jameswf | NADER/PAUL in 08 |
| 18:11.17 | Katty | Washy: if you watched the debates... |
| 18:11.28 | Katty | Washy: Ron Paul was usually the most worked up over things. |
| 18:11.29 | DonAlex | supjigator: Cause No. 90 - non-existent CUG.? |
| 18:11.30 | *** join/#asterisk esaym (n=user@cpe-72-183-198-134.satx.res.rr.com) |
| 18:11.33 | Katty | Washy: older, gray hair.. |
| 18:12.13 | Katty | Washy: I rather liked his position on a few things. |
| 18:12.35 | Washy | Pfft. They're all a bunch of liars |
| 18:12.51 | Katty | Washy: Sadly, the media had a big hand in keeping people from knowing his name or what he stood for. |
| 18:12.54 | Havokmon | I don't understand the whole primary thing |
| 18:12.55 | Katty | Washy: and other people too. |
| 18:13.13 | DonAlex | supjigator: Is that CUG as in Closed User Group then? Doesn't seem to make too much sense? |
| 18:13.17 | Washy | At least Nader is entertaining |
| 18:13.26 | Havokmon | Why, if these corporations -err political groups, are selecting their candidate, can I only vote for one? Why can't I select one from each group? |
| 18:14.06 | russellb | let's play a game called talk about asterisk |
| 18:14.09 | Washy | Havokmon: en.wikipedia.org/wiki/Approval_Voting |
| 18:14.14 | Katty | russellb: that gets boring. |
| 18:14.16 | russellb | politics discussion is not welcome |
| 18:14.22 | DonAlex | Oh crap yes it does.. |
| 18:14.26 | Katty | russellb: neither are you, but we tolerate you :P |
| 18:14.48 | Havokmon | yeah it has to be something Fender can jump down someone's throat about ;) |
| 18:14.48 | DonAlex | ok so Anyone know how to set the CUG for a C&W ISDN line in asterisk ? |
| 18:14.49 | russellb | my ban finger is warm today |
| 18:15.06 | Katty | russellb: i still love you. |
| 18:15.14 | tzanger | Katty: cable? |
| 18:15.15 | jameswf | bud dum crash |
| 18:15.21 | Katty | tzanger: :< |
| 18:15.25 | Katty | tzanger: i should call and pester them |
| 18:15.41 | Washy | TO get back on track, why would a normal user want to run an asterisk server? |
| 18:15.47 | *** join/#asterisk BobLutz (n=miles@d60-65-93-136.col.wideopenwest.com) |
| 18:15.51 | Washy | I only come here because you people know all about SIP |
| 18:15.55 | Washy | And I like that about you |
| 18:15.56 | jameswf | Washy: its the future |
| 18:16.01 | nvrpunk | Washy, free phone calls? |
| 18:16.12 | russellb | a super l33t answering machine? |
| 18:16.21 | Katty | cat feeder. |
| 18:16.28 | Katty | dial 20 to feed kittehs now |
| 18:16.33 | Washy | nvrpunk: BUt you need a PSTN provider right? |
| 18:16.33 | russellb | really fun torture you can unleash on unwanted calls? |
| 18:16.38 | Katty | dial 21 to eject CDrom drive |
| 18:16.49 | Katty | dial 22 to route all incoming calls to your extension to china |
| 18:16.56 | russellb | Washy: yup |
| 18:16.57 | nvrpunk | Washy, at some point, yes but my work has one |
| 18:17.07 | russellb | my work has one too, hehe |
| 18:17.16 | Washy | lol |
| 18:17.25 | nvrpunk | Washy, which means I can connnect to ir |
| 18:17.27 | nvrpunk | it* |
| 18:17.30 | lmadsen | nvrpunk: possibly cheaper phone calls, but no phone calls are free (at least not calls to the PSTN) |
| 18:17.35 | nvrpunk | and then use their long sdistance |
| 18:17.37 | Katty | dial 23 to call russellb and play monkey recording to him. |
| 18:17.40 | jameswf | someone needs to seriously revive the telephone torture stuff |
| 18:17.53 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 18:17.53 | lmadsen | I nominate jameswf |
| 18:17.56 | Katty | dial 24 to replace all audio files with BOH versions |
| 18:18.56 | Katty | if not sooner. |
| 18:19.04 | jameswf | 1st wife only lasted 5 heh |
| 18:19.12 | Katty | i wonder why. |
| 18:19.16 | jameswf | I am at 5 on this one |
| 18:19.17 | nvrpunk | also, I can create a voice prompt system for home use |
| 18:19.26 | nvrpunk | and setup inbound billing and invoice aholes i dont wish to talk to |
| 18:19.30 | nvrpunk | and mail them the invoice |
| 18:19.39 | Katty | nvrpunk: nvrpunk is not here. shut up and go away, or press 1 try him at work. |
| 18:19.51 | Washy | well thank you everybody. TOday has been fun |
| 18:20.08 | nvrpunk | 5's about the limit |
| 18:20.15 | Katty | i'm going to make a lolpbx. |
| 18:20.26 | Katty | for kitteh voicemailery press 2. |
| 18:20.54 | nvrpunk | Marriage - The triumph of imagination over intillect. |
| 18:20.54 | nvrpunk | intellect* |
| 18:21.04 | Washy | you guys are 1337. and I *heart* open VoIP software |
| 18:21.39 | keith4 | ... |
| 18:21.48 | russellb | jbot: hug Washy |
| 18:21.49 | jbot | ACTION hugs Washy |
| 18:21.58 | Washy | Yay! |
| 18:21.58 | Katty | jbot: good bot! |
| 18:21.58 | jbot | Katty: thanks |
| 18:22.05 | Washy | buh-bye |
| 18:22.05 | Katty | jbot: botsnack? |
| 18:22.05 | jbot | :), Katty |
| 18:22.09 | *** part/#asterisk Washy (i=Washy@gateway/tor/x-8465e4deaa2d9208) |
| 18:22.09 | jameswf | you know what they say if they told you that 60% of plains crashed you would never fly but 60% of marriges fail and people stand in line... wtf |
| 18:22.26 | jameswf | *planes |
| 18:22.36 | Katty | says the man that got remarried. |
| 18:22.40 | jameswf | sucker |
| 18:22.48 | Katty | congrats russellb! (= |
| 18:22.50 | seanbright | russellb: who's the luck guy? |
| 18:22.54 | russellb | thanks :) |
| 18:22.57 | jameswf | take it from me there wont be a 3rd |
| 18:22.57 | Katty | russellb: i wish you long and fullfilling tax refunds |
| 18:23.01 | Katty | russellb: and a good marriage too |
| 18:23.20 | *** join/#asterisk JunK-Y (n=junky@modemcable153.55-201-24.mc.videotron.ca) |
| 18:23.23 | Katty | YOU |
| 18:23.27 | Katty | JunK-Y: why i outta...just... |
| 18:23.30 | Katty | JunK-Y: HUG YOU! |
| 18:25.02 | jameswf | I got re-married cause I was sitting there one day and thought to my self WOW I have way to much money and more free time than any man deserves, how can I eliminate this situation |
| 18:25.10 | JunK-Y | whats up? |
| 18:25.19 | Katty | just bored. |
| 18:25.23 | nvrpunk | i get married soon |
| 18:25.47 | Katty | jameswf: you sound a little arrogant. |
| 18:25.55 | Katty | jameswf: your wife's a saint. |
| 18:26.11 | nvrpunk | code up a custom inbound billing system using php mysql and asterisk's cdr for me Katty |
| 18:26.12 | jameswf | damn I need to work on that cause I was going for SUPER arrogant |
| 18:26.19 | [TK]D-Fender | Katty: Nope,a gold-digger... a match made in heaven ;) |
| 18:26.21 | nvrpunk | k, thx |
| 18:26.28 | Katty | nvrpunk: already did. |
| 18:26.42 | Katty | nvrpunk: http://angela.sleekgeek.org/category/geekery/linux/asterisk/asterisk-stat/ |
| 18:26.45 | Katty | nvrpunk: go read. |
| 18:26.56 | jameswf | my wife is an angel .... yeah she is always up in the air harping about something... |
| 18:27.06 | nvrpunk | Katty, prepay or postpay? |
| 18:27.25 | Katty | nvrpunk: not for sale, sorry! |
| 18:27.58 | nvrpunk | Katty, i meant does it bill prepay or postpay |
| 18:28.12 | Katty | nvrpunk: you didn't read. |
| 18:29.29 | nvrpunk | ah |
| 18:29.49 | nvrpunk | Katty, that doesnt count as a billing solution :p |
| 18:30.10 | Katty | nvrpunk: ^_- |
| 18:30.17 | jameswf | I think A2Billing is the Defacto standard |
| 18:30.19 | Katty | nvrpunk: you meant ot tell me an accountant can't look at that and do some math? |
| 18:30.44 | nvrpunk | Katty, i was going for much more all inclusive |
| 18:30.51 | Katty | nvrpunk: ah right. |
| 18:31.02 | nvrpunk | Katty, we have 2000 end users and are ran by 2 people |
| 18:31.04 | Katty | nvrpunk: well maybe jameswf knows what's he's talking about (= |
| 18:31.07 | nvrpunk | so no |
| 18:31.31 | nvrpunk | and by ran by two people I mean one, because owner asks me to do stuff |
| 18:31.34 | nvrpunk | and I do it |
| 18:31.48 | Katty | the owner asks me to do stuff |
| 18:31.51 | Katty | and i tell him why it's a bad idea. |
| 18:31.55 | Katty | and then do it anyway |
| 18:32.32 | Katty | zugzug. |
| 18:33.15 | *** join/#asterisk jsmith (n=jsmith@72.21.36.138) |
| 18:33.15 | *** mode/#asterisk [+o jsmith] by ChanServ |
| 18:33.32 | Katty | Agent Smith |
| 18:34.23 | jsmith | Well, you know |
| 18:34.34 | jsmith | I gotta make my "every once in a while" appearance in the channel |
| 18:34.52 | [TK]D-Fender | jsmith: Annual quota achieved! |
| 18:35.17 | jsmith | [TK]D-Fender: Wow... lucky me. |
| 18:35.37 | jsmith | If you've got questions, throw them out, as I'm only gonna be around for a little while :-) |
| 18:35.52 | [TK]D-Fender | jsmith: No, lucky us..... to have succeeded in so much so soon into the year. |
| 18:35.52 | juanjoc | file: I've just created the ticket for the RTP timestamp skew. It's #12148. Just let me know if you want me to test anything. |
| 18:36.29 | jsmith | [TK]D-Fender: I ain't all that and a bag of chips |
| 18:37.25 | [TK]D-Fender | jsmith: Yes... chips are a billable "extra". |
| 18:37.43 | vap0rtranz | is there such a thing as a SIP channel heartbeat? |
| 18:38.07 | jsmith | vap0rtranz: Some devices will send a SIP OPTIONS packet to make sure the SIP peer is responding. |
| 18:38.18 | jsmith | vap0rtranz: Asterisk will do the same if you set "qualify=yes" |
| 18:38.57 | vap0rtranz | jsmith: qualify is good. i like the ms stats it gives. but does it include tx/rx that have to be torn down? |
| 18:39.20 | jsmith | vap0rtranz: I'm not sure I understand your question. |
| 18:39.30 | ThoMe | i search a graphic for "phone busy" :-) |
| 18:40.46 | vap0rtranz | jsmith: every minute or so, * boxes show several dozen channels with format (nothing) but i can't trace what's happening/if it's legit |
| 18:40.54 | *** join/#asterisk Nugget (i=nugget@carrera.macnugget.org) |
| 18:40.57 | *** join/#asterisk oej (n=olle@p5485E7EC.dip.t-dialin.net) |
| 18:41.14 | [TK]D-Fender | vap0rtranz: let me guess, you see this via "sip show channels"? |
| 18:41.32 | jsmith | vap0rtranz: Yes, that's probably the SIP OPTIONS packets that Asterisk is sending due to your "qualify=yes" |
| 18:41.37 | [TK]D-Fender | vap0rtranz: Those are not actual active calls, just misc SIP communications, etc. |
| 18:41.38 | vap0rtranz | [TK]D-Fender: yes, like [sip_ipaddy] [user] 354bad327b4 00117/00000 0x0 (nothing) No |
| 18:41.41 | x86 | is anyone able to convert this wav file to something usable by asterisk: http://www.wavsource.com/snds_2008-03-01_888350906694402/sfx/doorbell_x.wav |
| 18:41.56 | jsmith | vap0rtranz: You can turn on SIP history and see the dialog |
| 18:42.07 | Katty | Nugget: ! |
| 18:42.19 | [TK]D-Fender | vap0rtranz: Don't worry about all that... |
| 18:42.27 | *** join/#asterisk ZPertee (n=ZPertee@rrcs-24-106-241-121.central.biz.rr.com) |
| 18:42.27 | jameswf | x86 I bielieve if you google their are some online wav to gsm gonverters |
| 18:42.29 | jsmith | x86: The audio rate on that file is 22kHZ, instead of 8kHz |
| 18:42.41 | jsmith | x86: Use sox or audacity or something to convert it |
| 18:42.42 | vap0rtranz | jsmith: ok, so qualify should get rid of them. it just seems like the boxes i'm working with (1.2 & 1.4) sometimes get flooded with these and prevent dial-outs |
| 18:42.45 | *** join/#asterisk graphx (n=jeff@rrcs-67-79-144-150.se.biz.rr.com) |
| 18:42.49 | ZPertee | I have asterisk 1.4 how can I get flite working with it? |
| 18:43.06 | jsmith | vap0rtranz: They shouldn't prevent dial-outs... if they do, you've got bigger problems |
| 18:43.08 | x86 | jsmith: sox can't read it |
| 18:43.25 | jsmith | x86: Did you try audacity? You may have to tell sox the format it's in. |
| 18:43.30 | x86 | jameswf: i want it wav, preferrably, but it's in a format that sox and asterisk can not understand |
| 18:43.45 | x86 | jsmith: it's a wave file, but somehow it's using the mp3 codec |
| 18:43.58 | x86 | doorbell.wav: RIFF (little-endian) data, WAVE audio, MPEG Layer 3, mono 22050 Hz |
| 18:44.01 | jsmith | Yup... There are lots of different wav encodings |
| 18:44.03 | vap0rtranz | jsmith: yea i had hoped 1.4 would have fixed this. maybe it's Sysmaster's fault :( |
| 18:44.12 | redax | hi, |
| 18:44.13 | x86 | jsmith: audacity runs on a mac? |
| 18:44.19 | jsmith | x86: Not sure... |
| 18:44.42 | x86 | jsmith: tis why i was wanting some help converting it... not a lot out there for macs that can do it |
| 18:44.47 | redax | on debian linux, what's the best to do, use the stock debian kernel, or download the latest kernel.org and build my own? |
| 18:44.49 | jsmith | vap0rtranz: If you feel it's a bug in 1.4, feel free to open a bug on the bug tracker... but make sure you can document that it's the SIP OPTIONS packets that are causing the problem |
| 18:44.49 | x86 | none of my linux servers have X on them |
| 18:44.53 | Nugget | moo! |
| 18:45.11 | redax | the kernel question is for asterisk or course |
| 18:45.13 | graphx | Anyone very acquainted with Attended Call Transferring? |
| 18:45.14 | redax | of curse |
| 18:45.46 | [TK]D-Fender | graphx: Provide some details and see what you get. |
| 18:46.15 | vap0rtranz | jsmith: probably not a bug. Sysmaster just sucks. thx for the feedback |
| 18:47.09 | Yourname`` | jsmith: Is -users moderated? |
| 18:47.10 | graphx | sorry got a call |
| 18:47.14 | graphx | back now |
| 18:47.48 | jsmith | Yourname``: No, it's not moderated. But it does sit behind a fairly hefty (and sometimes over-active) spam filter. |
| 18:48.07 | jsmith | redax: It's up to you -- the stock debian kernel should be fine for most things |
| 18:48.22 | graphx | I am working with a Fedora Core installation of Asterisk and (in general) the blind transfers work at times and then they don't, the Attended Call Transfers don't work all around... |
| 18:48.41 | Yourname`` | jsmith: I just unsub'd as I tried two different emails and it didn't make it :S |
| 18:49.30 | jsmith | Yourname``: Send me a private message with your email address, and I'll help you out |
| 18:52.50 | graphx | I have gone through the documentation for Call Transferring and I am unable to find an exact document that covers this so I can learn to make changes to make the Attended Call Transfer work. I believe original it might have been working on default install. Without any good doc's on it I am unable to figure out what has change so I can fix it... |
| 18:53.45 | graphx | Any ideas of any documents I might be able to glean some infromation from to find a fix? |
| 18:54.05 | bsdwarrior | anyone around. im having problems with asterisk. the calls pickup , but my main menu isnt playing |
| 18:54.22 | bsdwarrior | when I show channels, I see the calls, but I call in and get dead air, the call is getting to asterisk |
| 18:55.17 | jsmith | bsdwarrior: Do you have any unconfigured T1 cards in the system? |
| 18:55.38 | jsmith | graphx: What, in detail, are you trying to figure out? |
| 18:55.41 | bsdwarrior | jsmith yes we had a problem with zaptel, so I commented out zapata.conf |
| 18:55.43 | Katty | does the name Shoretel ring a bell with anyone? |
| 18:56.10 | vap0rtranz | Katty: seen it, know nothing |
| 18:56.13 | jsmith | bsdwarrior: The problem is that you've got a T1 card that isn't taking interrupts, so everything is acting like it's working, but the audio doesn't actually get generated |
| 18:56.16 | Katty | hmm. |
| 18:56.23 | jsmith | Katty: Uh, yeah... |
| 18:56.24 | Katty | somehow, they got my name and number |
| 18:56.29 | Katty | and decided to pester me. |
| 18:56.38 | Katty | jsmith: details? |
| 18:56.39 | jsmith | Katty: I've used their stuff before.... :-( |
| 18:56.41 | bsdwarrior | jsmith the t1 is for outbound cards, i have pots lines fowarding to the asterisk box |
| 18:56.43 | Katty | i don't like sad faces. |
| 18:56.54 | Katty | jsmith: whyfor sad face. |
| 18:57.08 | jsmith | Katty: They had some nice parts and some bad parts... in my experience though, if you like Asterisk you won't be happy with ShoreTel |
| 18:57.21 | jsmith | Katty: We had some pretty serious problems with some of their hardware |
| 18:57.43 | Katty | jsmith: i like everything about asterisk except Pretty Software options and 24/7 tech support :/ |
| 18:57.47 | jsmith | Katty: And they tried to nickel and dime us to death on things like voicemail boxes, phone warranties, etc. |
| 18:57.58 | Katty | jsmith: hrmm. |
| 18:58.12 | Katty | jsmith: what about from an end user point of view. |
| 18:58.14 | Katty | jsmith: is it pretty? |
| 18:58.15 | x86 | jsmith: ok, got Audacity installed and the wav file loaded into it... how do i save it as a regular wav49 file with 8000Hz sample rate like Asterisk wants? |
| 18:58.26 | jsmith | Katty: Digium offers versions of Asterisk with a pretty face and 24/7 tech support ;-) |
| 18:58.48 | Katty | jsmith: not for us. for our clients |
| 18:58.57 | Katty | jsmith: i am tech support :< me. only me :< |
| 18:59.04 | Katty | :<<< |
| 18:59.05 | bsdwarrior | jsmiwth ideas? |
| 18:59.26 | jsmith | bsdwarrior: Configure the T1 cards... put a "span=" line in zaptel.conf, rerun "ztcfg -vv" and see if that helps. |
| 18:59.36 | twisted | I FUCING HATE C |
| 18:59.39 | [TK]D-Fender | graphx: What kind of phones are you using? |
| 18:59.45 | jsmith | Katty: Have you checked out the Switchvox product? |
| 18:59.46 | Katty | twisted: hi |
| 18:59.49 | twisted | BYE |
| 18:59.52 | Katty | twisted: bye. |
| 19:00.05 | Katty | jsmith: never heard of it. * based? |
| 19:00.17 | jsmith | Katty: Yeah... |
| 19:00.22 | *** join/#asterisk styelz (n=yoohoo@2001:388:f000:0:0:0:0:20b) |
| 19:00.27 | *** join/#asterisk shido6 (n=shido6@204.126.120.132) |
| 19:00.27 | Katty | jsmith: opinon? |
| 19:00.31 | jsmith | Katty: It's not a perfect fit for everyone, but it does have a slick interface |
| 19:00.41 | jsmith | Katty: You can download a free version and try it out |
| 19:00.59 | Katty | jsmith: server based, or blender. |
| 19:02.08 | *** join/#asterisk tld (n=terje@217-14-11-114-dhcp-osl.bbse.no) |
| 19:02.37 | tld | Is Asterisk happy with G722? |
| 19:02.48 | jsmith | tld: Define "happy" |
| 19:03.13 | tld | jsmith: Will it use it? And take advantage of it where doing so makes sense? Such as a call between two devices supporting it? |
| 19:03.15 | jsmith | tld: Asterisk will passthrough G722, and will even play prompts and record audio in G722. But I don't think it does transcoding of G722 yet. |
| 19:03.20 | *** join/#asterisk stansmith (n=stansmit@adsl-68-252-62-180.dsl.wotnoh.ameritech.net) |
| 19:03.20 | Katty | jsmith: looks like something similiar to Rhino Equipment. |
| 19:03.28 | tld | jsmith: Sounds like my needs are covered. Thanks. :) |
| 19:03.34 | jsmith | tld: You're welcome! |
| 19:04.05 | *** join/#asterisk docelmo (n=vircuser@h74.196.180.12.cable.rstb.cablerocket.net) |
| 19:04.20 | Katty | docelmo: HERRO |
| 19:04.31 | docelmo | MEW! |
| 19:04.35 | *** join/#asterisk gego (n=gego@host-091-097-120-132.ewe-ip-backbone.de) |
| 19:04.37 | BobLutz | heh |
| 19:04.41 | *** part/#asterisk stansmith (n=stansmit@adsl-68-252-62-180.dsl.wotnoh.ameritech.net) |
| 19:04.57 | docelmo | MEW MF'r Im a mad KATTY.. MEW MF'r.. :) |
| 19:05.09 | tzanger | hahahaha |
| 19:05.11 | Katty | no, not mad today. |
| 19:05.13 | tzanger | mew mf'er |
| 19:05.14 | tzanger | hahaha |
| 19:05.17 | Katty | but that could change in 5 minutes. |
| 19:05.30 | tzanger | that looks like something that needs a place on icanhazcheezburger |
| 19:05.34 | docelmo | and you are not even a fire crotch.. go figure.. |
| 19:05.58 | *** join/#asterisk actros1840 (n=A@chello087207097013.chello.pl) |
| 19:06.13 | robmac67 | jsmith: can you tell me how to specify a stun server in gtalk/jabber.conf ? |
| 19:06.15 | docelmo | so ANYWHO.. anyone know how many call files the manager can handle? I just sent mine about 1500 originate via the manager API and I got a shit load of these.. chan_sip.c:2918 auto_congest: Auto-congesting SIP/aretta-09072be8 |
| 19:06.38 | jsmith | robmac67: Nope... I have no clue on gtalk/jabber.conf |
| 19:06.46 | jsmith | robmac67: I haven't yet wrapped my head around it |
| 19:07.36 | robmac67 | jsmith: ok, thanks |
| 19:08.12 | errr | anyone using aastra 55i phones happen to know how to make text wrap to a new line if its to long to display on 1 ine? |
| 19:08.16 | errr | line* |
| 19:08.16 | Katty | locate xmms |
| 19:08.27 | jsmith | actros1840: Install format_mp3 from asterisk-addons |
| 19:08.33 | jsmith | actros1840: Then just use Playback |
| 19:08.48 | vap0rtranz | Katty: isn't xmms dead already?! |
| 19:08.59 | errr | xxms died for ever ago |
| 19:09.17 | errr | development wise |
| 19:12.03 | actros1840 | jsmith : but there is one big drawback for Playback and MP3Player, there is no way to gain the volume on demand |
| 19:13.57 | BobLutz | What is the biggest difference between Verbose() and NoOp()? |
| 19:14.17 | *** join/#asterisk k3mp (n=k3mp@pD9EBC004.dip.t-dialin.net) |
| 19:14.19 | jsmith | BobLutz: With Verbose(), you can control the verbosity level at which the text is displayed on the CLI |
| 19:14.29 | jsmith | BobLutz: With NoOp(), it's always verbosity of 3 or higher |
| 19:15.16 | BobLutz | Ahhhh! Thank you! |
| 19:15.34 | k3mp | Hello there, is anyone here who could help me proving that a called number ist disconnected? |
| 19:16.15 | graphx | I am using SPA900 series Linksys |
| 19:16.16 | k3mp | im always getting congestion back |
| 19:16.19 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
| 19:16.31 | *** join/#asterisk inv_arp (n=junya@c-76-26-24-197.hsd1.fl.comcast.net) |
| 19:16.33 | x86 | k3mp: call the LEC |
| 19:16.36 | *** join/#asterisk killfill (n=killfill@200.73.13.54) |
| 19:16.52 | killfill | hey guys.. im trying to force a restart on the zap channels.. |
| 19:17.00 | killfill | [Mar 5 16:16:05] ERROR[637]: chan_zap.c:9459 start_pri: Unable to open D-channel 16 (Device busy) |
| 19:17.01 | k3mp | x86: LEC? |
| 19:17.07 | killfill | what does this mean?... |
| 19:17.14 | killfill | busy in what sense?... noboady is using it... |
| 19:17.29 | killfill | how can i debug this? |
| 19:17.56 | jsmith | killfill: Stop Asterisk, restart Zaptel, and then restart Asterisk |
| 19:18.09 | graphx | jsmith, [TK]D-Fender I am on my way to a location. I will be back later today. Thanks for your time so far.. |
| 19:18.33 | *** join/#asterisk maniac777 (n=maniac@ppp-2-149.pppoe.sistbg.net) |
| 19:19.19 | killfill | :S |
| 19:20.41 | *** join/#asterisk RoyK (n=roy@158.36.187.198) |
| 19:21.31 | docelmo | anyone have any ideas on the auto-congestion issue win chan_sip? |
| 19:21.52 | *** join/#asterisk bluequijote (n=chatzill@user-10cm3kj.cable.mindspring.com) |
| 19:22.12 | killfill | this is wired.. |
| 19:22.26 | killfill | i see the alarmas OK with zap show status |
| 19:22.50 | killfill | but i cannot use ZAP |
| 19:23.12 | x86 | k3mp: LEC == Local Exchange Carrier |
| 19:23.19 | killfill | i dont see the typical mesage "starting channel X" |
| 19:23.30 | x86 | killfill: check your logs |
| 19:25.49 | killfill | [Mar 5 16:25:31] DEBUG[1403] chan_zap.c: Failed to read gains: Invalid argument |
| 19:26.03 | killfill | i see that message for all channels... |
| 19:27.49 | BobLutz | killfill: Seems like you have an invalid option set in one of your config files |
| 19:29.50 | AndyGraybeal | BobLutz: i know some lutz's in lancaster ohio |
| 19:30.34 | BobLutz | AndyGraybeal: Oh yeah? Are you from Lancaster? |
| 19:30.58 | AndyGraybeal | yea, sugar grove really |
| 19:31.54 | x86 | killfill: pastebin your zaptel.conf and zapata.conf |
| 19:32.49 | killfill | BobLutz: you mean in extensions.conf?... |
| 19:33.28 | AndyGraybeal | BobLutz: i've done computer work which for the police dept. there in lancaster |
| 19:34.19 | BobLutz | killfill: Most likely zapata.conf |
| 19:34.23 | x86 | killfill: no, he means in either zaptel.conf or zapata.conf |
| 19:34.44 | BobLutz | AndyGraybeal: What kind of work? |
| 19:34.58 | *** join/#asterisk CCFL_Man2 (i=ebb4cab4@pool-70-105-214-214.scr.east.verizon.net) |
| 19:34.59 | *** join/#asterisk troubled (n=troubled@pdpc/supporter/sustaining/troubled) |
| 19:35.05 | killfill | http://pastebin.ca/929421 |
| 19:35.10 | killfill | this are my zapata and zaptel.. |
| 19:35.20 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 19:37.05 | x86 | you need a context |
| 19:37.08 | *** join/#asterisk mike-ekim (n=mike@apoc.digiport.com) |
| 19:37.24 | mike-ekim | can I play mp3 file as an IVR msg? or does it have to be GSM. if so, can i convert mp3 to gsm? |
| 19:37.27 | Katty | you people sure whine a lot. |
| 19:37.33 | AndyGraybeal | BobLutz: ah networking stuff.. i don't even remember... ripping out old 2mbit network crap.. installing 10mbit.. it was a while ago |
| 19:37.51 | AndyGraybeal | gave'em a fileserver, workstations.. etc.. |
| 19:37.56 | killfill | x86: context=DID_span_1 |
| 19:37.57 | killfill | its there... |
| 19:38.32 | BobLutz | Must of been neat to see the inside of a police station while not incarcerated |
| 19:38.36 | *** join/#asterisk rsc (n=robert@fedora/rsc) |
| 19:38.43 | killfill | whats happening?!... |
| 19:38.50 | x86 | mike-ekim: it can be almost any format but mp3... gsm, wav, wav49, sln, ulaw, alaw, g726, g722, g729, g723, g728.... |
| 19:39.00 | x86 | killfill: give me ssh access? |
| 19:39.15 | killfill | ooh cannot the mashin is no in the internet.. :S |
| 19:39.32 | killfill | wired is that the alarm is OK... |
| 19:39.57 | *** join/#asterisk sakic (n=sakic@adsl-20-174-37.clt.bellsouth.net) |
| 19:40.12 | [TK]D-Fender | mike-ekim: If you installed asterisk-addons you can use MP3's |
| 19:40.13 | rsc | Good evening. Is there a possibility with app_voicemail and IMAP to move mails per user/extension into a special mailbox? Because the option imapfolder is only a general one, not a per-user/-extension one :( |
| 19:40.27 | x86 | killfill: zaptel is happy, asterisk is not |
| 19:40.34 | x86 | killfill: so the problem is in zapata.conf |
| 19:40.46 | x86 | killfill: take out the trunkgroups completely |
| 19:41.05 | sakic | I am getting an error: This phone (IP address 192.168.2.12) does not appear to be on the same network as the PBX (IP address 192.162.2.199). |
| 19:41.20 | sakic | where does the pbx address come from? the ip is 192.168.2.199 |
| 19:41.22 | Qwell | sakic: error from? |
| 19:41.29 | *** join/#asterisk Poincare (n=jefffnod@amp89.ampersant.be) |
| 19:41.37 | sakic | using switchvox to diagnose a phone that isn't working :( |
| 19:42.05 | [TK]D-Fender | sakic: remote phone behind its own NAT? |
| 19:42.17 | x86 | killfill: try this: http://pastebin.ca/929430 |
| 19:42.59 | x86 | killfill: also, what kind of phones are you using? you only want to use echo cancellation if you're going from digital to analog |
| 19:43.15 | x86 | killfill: aka zap to sip or so |
| 19:43.27 | killfill | i just care about zap now.. :( |
| 19:43.57 | x86 | then don't use echo cancellation either |
| 19:44.00 | killfill | x86, loos this is the full log of zap restart. http://pastebin.ca/929431 |
| 19:44.05 | x86 | comment those two lines out |
| 19:44.21 | x86 | ah |
| 19:44.26 | x86 | take out the rxgain and txgain |
| 19:45.20 | sakic | TK - it is behind a router, the same one as the server |
| 19:45.42 | sakic | new system, one phone is working fine, one is not |
| 19:45.47 | [TK]D-Fender | sakic: is the phone attempting to connect to * via the WAN address? |
| 19:46.20 | hmmhesays | compiling compiling func_odbc, cause it is my friend for my directory |
| 19:46.31 | killfill | x86, ok i took tx and rx gain out, and the trunkgroup. same thing.. no zap.. :S |
| 19:46.33 | Katty | hmmhesays: hi! |
| 19:46.37 | Katty | hmmhesays: i got asterisk-stat working (= |
| 19:46.43 | Katty | hmmhesays: |
| 19:46.55 | Katty | hmmhesays: http://angela.sleekgeek.org/category/geekery/linux/asterisk/asterisk-stat/ |
| 19:47.00 | hmmhesays | Katty: awesome sorry I had to ditch you on that |
| 19:47.04 | killfill | damed.. it was working until today. |
| 19:47.07 | hmmhesays | I had some crazy stuff come up with callweaver |
| 19:47.16 | Katty | hmmhesays: all good. |
| 19:47.24 | Katty | hmmhesays: i have friends ;) |
| 19:47.50 | k3mp | Could someone help me with SIP_HEADER() ? |
| 19:47.57 | Katty | k3mp: for what porpoise |
| 19:47.58 | hmmhesays | :) nice, that is a fancy looking blog thing you have there |
| 19:48.12 | killfill | hm.. [Mar 5 16:48:00] ERROR[1679] chan_zap.c: Write to 52 failed: Protocol error |
| 19:48.18 | killfill | whats is this?... |
| 19:48.21 | Katty | hmmhesays: yeah, or else i forget how to do stuff. |
| 19:48.27 | Katty | hmmhesays: it's like tagging my memories. |
| 19:48.30 | Katty | hmmhesays: very handy |
| 19:48.31 | k3mp | Katty: i need to verify the code i get back |
| 19:48.39 | hmmhesays | Katty I use a personal version of mediawiki for that |
| 19:48.45 | hmmhesays | www.thelostpacket.org |
| 19:48.55 | Katty | k3mp: what, precisely, are you trying to accomplish? |
| 19:49.05 | sakic | TK not sure, we didn't configure the phone, it is using dhcp |
| 19:49.32 | k3mp | Katty: i'd like to update a database, when i get error 404 back |
| 19:49.34 | Katty | hmmhesays: that's very neat. |
| 19:49.35 | [TK]D-Fender | sakic: DHCP doesn't say what IP to connect to * on.... |
| 19:49.52 | sakic | do I check it out via the web address of the phone? |
| 19:49.53 | hmmhesays | same idea different format |
| 19:50.05 | killfill | http://pastebin.ca/929440 this are my protocol errors.. :S |
| 19:50.11 | Katty | k3mp: i'm sorry. i don't understand what you're trying to tell me. |
| 19:50.20 | Katty | k3mp: what type of phones are you using? Polycom? |
| 19:50.20 | [TK]D-Fender | sakic: You have no idea how to admin your own phones? |
| 19:50.37 | sakic | all mine are set up, this is a buddy's |
| 19:50.44 | sakic | he made some sort of mistake |
| 19:50.57 | k3mp | Katty: if im doing a SIP call to a disconnected number, i get error 404 back, but i need to to a database update then |
| 19:50.59 | sakic | and I am trying to help him see where the issue is |
| 19:51.09 | [TK]D-Fender | sakic: Go figure out how the phones are configured |
| 19:51.10 | *** join/#asterisk spatulamaan (n=ggilmore@207.188.8.252) |
| 19:51.11 | Katty | k3mp: oh. i see. |
| 19:51.35 | k3mp | Katty: to safe the moment when i proved this number to be invalid |
| 19:51.46 | Katty | k3mp: the only thing I know how to do with SIP Headers is to do polycom auto answer or ring answer. |
| 19:51.54 | Katty | k3mp: i don't think that will be able to help you any. |
| 19:52.12 | Katty | k3mp: i can link you the post, if you want it. |
| 19:52.18 | sakic | uh thanks? |
| 19:52.31 | *** join/#asterisk vader-- (n=me@c-71-226-192-99.hsd1.nj.comcast.net) |
| 19:52.33 | vader-- | hello |
| 19:52.42 | Katty | sakic: go check your phones, [TK]D-Fender thinks that's where your problem lies. |
| 19:52.44 | k3mp | Katty: yes, i'd like to have a look |
| 19:52.52 | *** join/#asterisk eric2 (n=nobody@sudbury-cable-69-60-240-185.unitz.ca) |
| 19:52.57 | vader-- | could anyone help me out i am trying to make an international call to +49 (0) 6123 7913-0 |
| 19:53.07 | vader-- | and my dial plan for international calls doesn't seem to work |
| 19:53.15 | hmmhesays | vader--, show it |
| 19:53.30 | Katty | k3mp: http://angela.sleekgeek.org/2007/11/06/freepbx-paging-and-intercom/ |
| 19:54.01 | Katty | k3mp: number 5 shows a snippet of code that might help you |
| 19:54.04 | vader-- | this is my dial plan line |
| 19:54.41 | vader-- | exten => _9011XXXXXXXXXXXX,n,Dial(${TRUNK}/${EXTEN:!}) |
| 19:55.01 | killfill | x86: what else can i try?.. :) |
| 19:55.09 | vader-- | when i dial i am dialing 9011490612379130 |
| 19:55.57 | vader-- | i just get a busy signal |
| 19:56.04 | k3mp | Katty: I found it, thanks |
| 19:56.05 | jsmith | vader--: I think you need to drop the 0 in parentheses |
| 19:56.17 | k3mp | Katty: I found it, thanks4help |
| 19:56.57 | vader-- | i get a progress with cause code 41 received |
| 19:57.00 | vader-- | in the console |
| 19:58.45 | Katty | k3mp: hope it helps you out (= |
| 19:59.01 | killfill | damn.. what does this mean? [Mar 5 16:58:10] DEBUG[1756] chan_zap.c: Failed to read gains: Invalid argument |
| 19:59.24 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 20:00.07 | k3mp | Katty: well id actually does not.. :| |
| 20:02.16 | [TK]D-Fender | vader--: And what is it you think ${EXTEN:!} is going to give you? You need to stop holding "shift" the same way I need to stop leaving on "caps-lock" |
| 20:02.27 | outtolunc | !not <G> |
| 20:04.14 | *** join/#asterisk dlynes (n=dlynes@mail.247communications.com) |
| 20:04.47 | dlynes | Anyone on atm that offers decent quality toll free numbers that are accessible from anywhere in continental US and Canada? |
| 20:05.14 | k3mp | Katty: i found a interesting text, which is about my problem |
| 20:05.17 | k3mp | Katty: http://www.voip-info.org/wiki/index.php?page=Asterisk%20variable%20DIALSTATUS |
| 20:05.54 | *** join/#asterisk Pagautas (n=bigman@ns.voip.ktu.lt) |
| 20:06.09 | dlynes | I need it to be able to work at minimum from Indiana, Illinois, Missouri, Georgia, British Columbia, and Ontario |
| 20:08.09 | [TK]D-Fender | dlynes: Bell Canada |
| 20:08.47 | dlynes | [TK]D-Fender: Anyone cheaper? |
| 20:09.02 | dlynes | [TK]D-Fender: I've already got GT/Bell |
| 20:09.25 | [TK]D-Fender | dlynes: I use Allstream personally. |
| 20:09.25 | dlynes | [TK]D-Fender: They're 7c/min, no connection charges, no signup charge |
| 20:09.51 | [TK]D-Fender | dlynes: .07$? $$# crazy. I get $.03 and didn't even really fight for anything |
| 20:10.14 | dlynes | [TK]D-Fender: yeah...gt/bell sucks...they know they're the heavyweight, and they push that weight for all it's worth |
| 20:10.36 | [TK]D-Fender | dlynes: just call up Allstream and they'll terminate to your lines. |
| 20:10.49 | dlynes | [TK]D-Fender: ok |
| 20:11.00 | dlynes | [TK]D-Fender: any setup charge, or connection charges for that 3c/min? |
| 20:11.10 | [TK]D-Fender | dlynes: Nope. |
| 20:11.15 | dlynes | [TK]D-Fender: ok, thansk |
| 20:11.27 | [TK]D-Fender | dlynes: Np. Mind you I've been witht hem for all my lines for years... |
| 20:11.46 | [TK]D-Fender | dlynes: And I'm preparing a win-back bidding war to maybe go back to Bell. |
| 20:12.11 | husimon | hey how do I turn on verbose logging for skinny? |
| 20:12.16 | husimon | I am getting messages like this : [Mar 5 10:11:37] WARNING[23865]: chan_skinny.c:4168 handle_message: Client sent message #0 without first registering. |
| 20:12.19 | husimon | and I don't know what host it's form |
| 20:12.21 | dlynes | [TK]D-Fender: ok, thanks |
| 20:12.22 | husimon | form=from |
| 20:12.23 | dlynes | [TK]D-Fender: gotta run |
| 20:12.38 | husimon | nm |
| 20:13.07 | husimon | i was using the wrong syntax |
| 20:13.38 | husimon | crap it still won't tell me what client is causing that error |
| 20:13.45 | husimon | any ideas [TK]D-Fender ? |
| 20:13.49 | *** join/#asterisk l0verb0y (n=l0verb0y@119.111.96.126) |
| 20:17.49 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 20:19.51 | *** join/#asterisk bkw__ (n=brian@adsl-75-60-114-157.dsl.tul2ok.sbcglobal.net) |
| 20:20.00 | docelmo | [TK]D-Fender ever seen chan_sip throw up this error: auto_congest: Auto-congesting SIP/XXXXX-0ae9dd38 |
| 20:20.33 | *** join/#asterisk lanning (n=lanning@ip67-152-85-190.z85-152-67.customer.algx.net) |
| 20:20.43 | docelmo | or anyone for this matter. And if so.. WHAT the hell causes it/ |
| 20:20.44 | docelmo | ? |
| 20:22.12 | *** join/#asterisk Greek-Boy (n=email@41.221.58.4) |
| 20:25.01 | *** join/#asterisk asteriskmonkey (n=asterisk@69.77.169.14) |
| 20:25.46 | *** join/#asterisk mecredis (n=fcb@cpe-74-73-135-105.nyc.res.rr.com) |
| 20:25.50 | *** join/#asterisk mchou (n=mchou@c-71-198-127-234.hsd1.ca.comcast.net) |
| 20:25.52 | *** join/#asterisk atis_home (n=chatzill@193.238.213.215) |
| 20:25.52 | mecredis | hi there |
| 20:25.56 | asteriskmonkey | how can you use chan/local and keep caller id? I know the __ with a variable makes in inherited permanently through out that call, but how do you set callerid with a __ ? |
| 20:26.01 | husimon | docelmo, http://bugs.digium.com/view.php?id=765 |
| 20:26.05 | husimon | that might be it |
| 20:26.08 | mecredis | I'm having issues with setting up a Sipura 1001 |
| 20:26.18 | mecredis | basically I need the unit to drop my handset into a dialplan |
| 20:26.29 | mecredis | I can dial the sipura |
| 20:26.42 | mecredis | but nothing happens after I push buttons at the dialtone |
| 20:26.45 | mecredis | on the handset |
| 20:26.51 | Katty | In terms of Chanspy, what's the difference between 'w' for whisper and 'W' for private whisper? |
| 20:26.57 | Katty | I don't really get the difference. |
| 20:27.12 | mecredis | anyone have any experience here? |
| 20:27.14 | husimon | might be who get's the whisper |
| 20:27.42 | Katty | hmm. |
| 20:27.44 | Katty | could be. |
| 20:27.45 | husimon | Katty, it says what it means in the wiki |
| 20:27.46 | husimon | w Enable 'whisper' mode, so the spying channel can talk to the spied-on channel. |
| 20:27.46 | husimon | - W Enable 'private whisper' mode, so the spying channel can talk to the spied-on channel but cannot listen to that channel. |
| 20:27.47 | mchou | I've got a general question regarding SIP and firewall nat. If I have multiple sip clients running behind my firewall, do they all need to have different SIP ports if they connect to the same SIP server? |
| 20:28.00 | Katty | yes, husimon, it does, but i don't understand what that means. |
| 20:28.04 | *** join/#asterisk atis_home (n=chatzill@193.238.213.215) |
| 20:28.22 | *** join/#asterisk FarrisG (n=jrush@gateway.wiquest.com) |
| 20:28.30 | Katty | anyone use chanspy with w and W options? |
| 20:28.40 | husimon | well the first one means the spying person can talk on the spied channel |
| 20:28.52 | husimon | so if I spy on you I can say BOO and you'll hear it |
| 20:29.04 | Katty | and the other person i'm talking to as well? (SIPB) |
| 20:29.20 | husimon | the second one means you can't hear the spied channel but they can hear you |
| 20:29.32 | Katty | that does not parse. |
| 20:29.33 | husimon | i guess that's more for like intercom stuff |
| 20:29.47 | husimon | like i'm cutting into your call to make an announcement but I dont' want to hear you |
| 20:29.53 | Katty | ah |
| 20:29.56 | Katty | hrmm. |
| 20:30.05 | husimon | that's what the english sentence means to me, someone correct me if i'm wrong. |
| 20:30.09 | Katty | so when using 'w', can both SIPA and SIPB hear you? |
| 20:30.11 | asteriskmonkey | if im using chan local how can i set caller id without loosing it |
| 20:30.31 | Katty | or just SIPA, who you chanspied on. |
| 20:30.59 | husimon | i think the channel probably means both of them |
| 20:31.12 | Katty | k |
| 20:31.28 | husimon | i'm not sure about that though |
| 20:31.32 | husimon | you should test it |
| 20:31.53 | Katty | yeah.. |
| 20:31.54 | FarrisG | I just upgraded to 1.2.26.1. Everything's working fine except zap, which I only use for a door phone. It's setup as a "batphone", with one channel a TDM400P. Module loads fine, and it will actually dial, but there is no audio on either end once it connects. zttool shows the device is ok |
| 20:32.01 | Katty | it would require me putting another phone on this server... |
| 20:32.07 | Katty | only two on there right now |
| 20:32.10 | Katty | but still doable. |
| 20:32.17 | husimon | just use a softphone or something |
| 20:32.20 | Katty | meh |
| 20:32.41 | Katty | i'd rather do a REAL test (= |
| 20:32.50 | husimon | that is a real test |
| 20:32.57 | mike-ekim | can I play mp3 file as an IVR msg? or does it have to be GSM. if so, can i convert mp3 to gsm? |
| 20:34.02 | [TK]D-Fender | mike-ekim: I already answered you. |
| 20:34.16 | [TK]D-Fender | mike-ekim: If you installed asterisk-addons you can use MP3's <---------- |
| 20:34.42 | docelmo | husimon I looksed at it. I changed the value higher. Um, I dont know if this will fix my problem or not. I load the manager API up with about 1500 calls and get that crap all over the place |
| 20:35.39 | *** join/#asterisk hi365_m (n=hi365@213.151.53.95) |
| 20:37.14 | husimon | docelmo, yeah i'm not really sure I just googled the error. |
| 20:37.35 | bsdwarrior | usually when I type reload, it fixes our "echo" issues. any other suggestions. the config has always worked |
| 20:38.43 | *** join/#asterisk CrashSys (i=0@rrcs-71-41-32-190.se.biz.rr.com) |
| 20:38.43 | hi365_m | anyone got a good source for * hardware in the uk? |
| 20:38.58 | docelmo | I will do the same.. Like I siad.. I send about 1500 calls via manager originate API and it craps everywhere with the auto-congest deal.. |
| 20:39.22 | *** join/#asterisk ELBunce (n=erik@kde/developer/bunce) |
| 20:39.43 | jameswf | hi365_m: Voipon |
| 20:39.48 | CrashSys | Anyone know how I can get Asterisk to give me useful output from the monitor command? Trying to run a script at MONITOR_EXEC and it doesn't do anything, or doese it not run immediately? |
| 20:40.40 | hi365_m | jameswf: thanks. how are the prices compared to the local amrket? |
| 20:40.56 | *** join/#asterisk Jicksta (n=jicksta@dsl093-128-144.sfo4.dsl.speakeasy.net) |
| 20:41.10 | docelmo | CrashSys read the docs or enable debug on your console CLI.. |
| 20:41.16 | docelmo | or read the wiki |
| 20:41.29 | jameswf | well they should be about the same usualy the prics are set just a matter of how they translate in given currency |
| 20:41.40 | CrashSys | I'd like to enable debug on the console, but I get a few hundred sip debug messages every few seconds |
| 20:41.54 | CrashSys | read the wiki... |
| 20:42.03 | docelmo | ~wiki |
| 20:42.13 | CrashSys | I've already read the wiki |
| 20:42.45 | jameswf | hi365_m: site is http://www.voipon.co.uk/ you can check their prices |
| 20:44.12 | Kobaz | i make a call from a sip phone, and dial out onto the pstn. When connecting to certain destinations (especially cell phone voicemail), i will miss the first 5 seconds of the audio stream... anyone know how to fix this? |
| 20:44.18 | AndyGraybeal | BobLutz: yea, it was interesting... last time i was with the cops i was behind the bars.. |
| 20:44.34 | bsdwarrior | calls are ringing to a phone when it already has a call. sip prune realtime user ext and sip prune realtime peer ext doesnt work when they are on the call |
| 20:44.41 | AndyGraybeal | BobLutz: they even let me work over in the undercover narcotics section |
| 20:45.46 | fujin | anyone have a script to convert voicemail.conf into realtime sql? |
| 20:45.51 | fujin | or do I have to write another one |
| 20:46.37 | CrashSys | So anyways, i've got "MONITOR_EXEC=/usr/local/bin/ast2mp3" set in the global's section. Debug doesn't show it trying to run the script. I'm calling monitor with Monitor(wav,${file},mb) |
| 20:46.41 | fujin | agh, this pains me |
| 20:48.01 | CrashSys | Any idea how I can get Asterisk to tell me what it's doing when it thinks it's running the script? Or if it even is? (it's no in the wiki) |
| 20:48.12 | CrashSys | debug = 10 |
| 20:48.15 | CrashSys | verbose = 10 |
| 20:48.44 | mecredis | does anyone have experience with Sipura units? |
| 20:48.46 | Katty | anyone know how to do a whisper page, so only SIPA hears you, and not the third party? |
| 20:48.53 | mecredis | is there a way I can skip the dial tone? |
| 20:49.31 | husimon | skip it? like don't hear it? |
| 20:49.51 | mecredis | basically I want to pick u pthe phone |
| 20:49.56 | mecredis | and have it connect me automatically |
| 20:50.06 | mecredis | w/o having to dial a digit |
| 20:50.48 | [TK]D-Fender | mecredis: Depends on the phone. |
| 20:50.50 | mecredis | the sipura / SIP seems to require me to put a digit in |
| 20:50.58 | mecredis | [TK]D-Fender: the phone or the sipura unit? |
| 20:51.16 | mecredis | [TK]D-Fender: if you're thinking I can auto-program the phone to drop a digit in then I can't do that on this handset |
| 20:51.32 | [TK]D-Fender | mecredis: In this case the SPA. Yes there is an option for this. Go check out the forums on www.voxilla.com for instructions on how to do this. |
| 20:51.48 | mecredis | any particular place on there? |
| 20:52.17 | [TK]D-Fender | mecredis: the forums.... I'm sure you'll figure it out... |
| 20:52.19 | husimon | Katty, you know I found another link to the chanspy |
| 20:52.20 | husimon | <PROTECTED> |
| 20:52.20 | husimon | <PROTECTED> |
| 20:52.25 | mecredis | OK |
| 20:52.33 | husimon | i think w is actually what you want |
| 20:53.02 | husimon | that seems to me to say that the remote user won't hear but the local will |
| 20:53.42 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 20:53.54 | Katty | husimon: no, it's not. |
| 20:54.04 | Katty | husimon: we just tested it. both party's can hear it... it's just very faint and quiet. |
| 20:54.11 | Katty | husimon: i would consider 'w' Barge In |
| 20:54.25 | Katty | husimon: but a very /quiet/ barge in. I think what I'm actually wanting is Extenspy. testing it now (= |
| 20:54.34 | husimon | ah |
| 20:54.51 | husimon | guess that help page is wrong |
| 20:56.04 | vap0rtranz | i have 160 sip channels and growing!!! *EEK* |
| 20:56.22 | vap0rtranz | only a restart fixes this problem |
| 20:56.30 | jsmith | vap0rtranz: What version of Asterisk? |
| 20:56.38 | vap0rtranz | 1.4 |
| 20:56.43 | Kobaz | i make a call from a sip/iax phone, and dial out onto the pstn. When connecting to certain destinations (especially cell phone voicemail), i will miss the first 5 seconds of the audio stream... anyone know how to fix this? |
| 20:57.18 | ZaVoid | hey vap0rtranz i've seen that a lot too |
| 20:57.23 | vap0rtranz | jsmith: there's only 12 channels registered, so i don't see how this is related to qualify on each of those |
| 20:57.26 | ZaVoid | they all say like ACK at the end or somthing right? |
| 20:57.53 | vap0rtranz | ZaVoid: do you loose out/in ability? my end-users loose the ability to do most anything that's not internal |
| 20:58.11 | jsmith | vap0rtranz: If for some reason Asterisk isn't correctly destroying the dialog it creates when it sends the SIP OPTIONS packet, then they would stack up |
| 20:58.23 | vap0rtranz | jsmith: (200 channels) |
| 20:58.23 | *** join/#asterisk JerJer (n=PhatJ@pdpc/supporter/bronze/jerjer) |
| 20:58.31 | ZaVoid | ahh nope vap0rtranz sorry |
| 20:58.41 | vap0rtranz | jsmith: at this rate, the system will run out of resources eventually |
| 20:58.43 | ZaVoid | ahh is that why jsmith ? |
| 20:58.56 | ZaVoid | but i got qualify=no on all them come to think of it... |
| 20:58.57 | jsmith | vap0rtranz: What version of Asterisk? There were some problems early in the 1.4 series where the dialogs wouldn't get destroyed properly. |
| 20:59.46 | ZaVoid | mine are mostly (d) Rx: BYE actually at the end |
| 21:00.01 | vap0rtranz | jsmith: 1.4.17; just ported stuff over from <1.2.12 |
| 21:00.41 | jsmith | vap0rtranz: You may want to try the 1.4.19 release candidate, and see if that helps. |
| 21:00.53 | jsmith | vap0rtranz: There have been quite a few issues fixed since 1.4.17 was released |
| 21:00.57 | vap0rtranz | jsmith: diabling qualify incase it's the problem |
| 21:00.58 | *** join/#asterisk Nasra (n=maxshipp@CPE001217b1920e-CM00111ade9528.cpe.net.cable.rogers.com) |
| 21:01.34 | *** join/#asterisk oej (n=olle@p5485E7EC.dip.t-dialin.net) |
| 21:02.06 | *** join/#asterisk r0d3nt (i=nobody@69.31.131.44) |
| 21:03.00 | vap0rtranz | a restart always fixes this; i'll be looking back to trace when/how it started |
| 21:03.20 | *** join/#asterisk chodorenko (n=chodoren@ext.one.by) |
| 21:03.24 | chodorenko | Hi All |
| 21:03.59 | Katty | husimon: looks like Chanspy with w and Extenspy with w both accomplish the same thing. |
| 21:04.09 | Katty | husimon: two way audio with the extension i called. |
| 21:04.15 | Katty | husimon: third party can't hear them, at all |
| 21:04.18 | chodorenko | Asterisk supported cisco wildcard patterns in dial peer ? |
| 21:04.30 | Katty | husimon: do you, by chance, know how to do a barge in, where the third party CAN hear you? |
| 21:05.27 | husimon | Katty, I pretty much know as much as you do via reading the voip info page, I've never done any chanspy stuff before personally. |
| 21:05.30 | chodorenko | i can problem connect cisco to Asterisk by http digest SIP |
| 21:05.41 | husimon | I'd help you research but I'm busy fixing something atm. |
| 21:05.45 | Katty | husimon: ah, right. k |
| 21:05.53 | bsdwarrior | when I first start asterisk I have to type reload otherwise phones have bad echos . any ideas? |
| 21:06.49 | Katty | [TK]D-Fender: mew? |
| 21:07.21 | Kobaz | hmm, so noone knows why the beginning of a call might get chopped off? |
| 21:07.42 | chodorenko | FXS poert connect no probmlem , FXO ports have destination like 2020T auth in asterisk as sip:2020*@ip_of server |
| 21:07.59 | chodorenko | and not success autorize |
| 21:08.13 | [TK]D-Fender | Katty: nope. |
| 21:10.33 | Katty | [TK]D-Fender: k |
| 21:12.03 | vap0rtranz | jsmith: the logs make it look like qualify did do this but it's only because i have a problem with registration. all channels time out and qualify still keeps pilling on more "calls" |
| 21:13.20 | jsmith | vap0rtranz: Registration and qualify are totally independent from each other. |
| 21:13.40 | jsmith | vap0rtranz: The only thing I could possible think of is that maybe you're running out of file handles |
| 21:13.46 | jsmith | vap0rtranz: But that's a wild guess |
| 21:14.04 | vap0rtranz | jsmith: i'm not saying it's causal. i'm saying the kinds of calls would not have pilled up if the channels never timed out |
| 21:14.20 | *** join/#asterisk VaNNi (n=VaNNi@216.70.165.200) |
| 21:15.49 | *** join/#asterisk pithen (n=pithen@mail.graphlogic.com) |
| 21:16.08 | pithen | hey all, me again.. |
| 21:17.06 | pithen | is there any simple way to determine whether or not a call is local within the dialplan? Right now I'm basically making a hash map of all local area code+prefixes and using that to determine how to route the call..its a pita though |
| 21:17.36 | [TK]D-Fender | pithen: Yes its possible... if you complete your list. Keep at it |
| 21:17.50 | *** join/#asterisk RoyK (n=roy@ip-216-4-149-91.dialup.ice.no) |
| 21:18.09 | pithen | [TK]D-Fender, ahh, so thats basically the method you'd recommend? |
| 21:19.26 | [TK]D-Fender | pithen: thats basically all you can do. |
| 21:20.20 | pithen | btw, i think i've settled on getting one of the astribank units.. seems like the most cost effective, and pretty simple to set up |
| 21:20.28 | husimon | Unless someone already has a service that lets you query whether an area code is local |
| 21:20.31 | husimon | and then use some api to do it |
| 21:20.32 | husimon | heh |
| 21:20.57 | mike-ekim | .quit |
| 21:21.04 | pithen | husimon, true, but if it introduces any lag into the dialing ill have hell to pay ;) |
| 21:21.19 | *** join/#asterisk oej (n=olle@p5485E7EC.dip.t-dialin.net) |
| 21:21.19 | husimon | pithen yeah then your static map is best |
| 21:21.29 | husimon | there can't be _THAT_ many local area codes can there? |
| 21:21.45 | RoyK | does the g.729 codec support PLC yet? |
| 21:21.46 | Qwell | local prefixes |
| 21:21.59 | husimon | err ya |
| 21:22.11 | pithen | only two area codes, but not all prefixes are local...i've got 202 in my list |
| 21:22.19 | husimon | i guess if you live in nyc that's lame |
| 21:22.22 | Katty | [TK]D-Fender: do you have a url for a sort of barge in meetme example? |
| 21:22.37 | husimon | pithen this is to route calls via a free long distance? |
| 21:22.49 | Katty | [TK]D-Fender: SIPA is on phone with ZAP1, SIPB wants to do a conference call barge in bit and talk to both parties. |
| 21:23.05 | Katty | [TK]D-Fender: i can't find a variation of Chanspy or Extenspy to do that |
| 21:23.05 | pithen | husimon, not free.. i use asterlink for my long distance.. cheaper than telco |
| 21:23.07 | Corydon76-dig | ------------------------------------5 |
| 21:23.36 | Qwell | Corydon76-dig: wrong window ;D |
| 21:24.08 | pithen | although id like to integrate enum just in case theres a free route.. hopefully 1.4 is faster than 1.2. when i implemented it in 1.2 it took so long to dial that i finally disabled it |
| 21:26.45 | Qwell | http://yro.slashdot.org/article.pl?sid=08/03/05/1949226 |
| 21:26.46 | Qwell | pwnt |
| 21:29.34 | BobLutz | Hehe |
| 21:34.10 | *** join/#asterisk jcanfield (n=jcanfiel@68.109.242.162) |
| 21:35.17 | jcanfield | Anyone here ever networked asterisk to a Panasonic 576 using PRI? |
| 21:38.24 | *** join/#asterisk BCS-Satori (n=somewher@75.148.21.113) |
| 21:38.48 | BCS-Satori | How do i set asterisk only accept g711 u-law only? |
| 21:39.30 | [TK]D-Fender | BCS-Satori: "disallow=all" , "allow=ulaw" |
| 21:39.58 | BCS-Satori | Thank you |
| 21:41.33 | *** join/#asterisk Dovid (n=Dovid@bzq-79-180-58-141.red.bezeqint.net) |
| 21:41.40 | Dovid | anyone here using h323 on asterisk ? |
| 21:43.42 | tzafrir | and survived? |
| 21:44.23 | *** topic/#asterisk by russellb -> Asterisk: The Open Source PBX and Telephony Application Platform (asterisk.org) -=- Asterisk 1.6.0-beta5 (2008/03/05), Asterisk 1.4.19-rc1 (2008/03/05), *-Addons 1.4.6, 1.6.0-beta2 (2008/02/21), Zaptel 1.4.9.2 (2008/02/28), Libpri 1.4.3 (2007/12/13) -=- #asterisknow or #asterisk-gui for AsteriskNOW (asterisknow.org) -=- #switchvox for Switchvox (switchvox.com) -=- #freepbx for FreePBX/trixbox -=- FYI: sharks sighted to |
| 21:44.39 | *** join/#asterisk Jason99 (n=jason@jason.unitz.ca) |
| 21:44.51 | *** topic/#asterisk by russellb -> Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.0-beta5 (2008/03/05), Asterisk 1.4.19-rc1 (2008/03/05), *-Addons 1.4.6, 1.6.0-beta2 (2008/02/21), Zaptel 1.4.9.2 (2008/02/28), Libpri 1.4.3 (2007/12/13) -=- #asterisknow or #asterisk-gui for AsteriskNOW (asterisknow.org) -=- #switchvox for Switchvox (switchvox.com) -=- #freepbx for FreePBX/trixbox -=- FYI: sharks sighted today |
| 21:44.51 | Jason99 | Does anyone know if the meaning of nat=yes has been changed from 1.2 to 1.4 ? |
| 21:45.35 | *** join/#asterisk Pagautas (n=bigman@ns.voip.ktu.lt) |
| 21:45.50 | Dovid | haha |
| 21:46.00 | Dovid | tzafrir: I am trying to figure out which version to use |
| 21:47.51 | *** part/#asterisk lirakis (i=lirakis@66.252.24.133) |
| 21:48.28 | *** join/#asterisk ZPertee (n=ZPertee@rrcs-24-106-241-121.central.biz.rr.com) |
| 21:49.02 | *** part/#asterisk mecredis (n=fcb@cpe-74-73-135-105.nyc.res.rr.com) |
| 21:49.10 | ZPertee | can I install asterisk before I install zaptel? |
| 21:49.33 | jsmith | Yes, but you won't get any zaptel functionality |
| 21:49.54 | jsmith | Asterisk detects whether or not Zaptel is installed, and if it doesn't find it, it doesn't compile the chan_zap module |
| 21:50.11 | jsmith | Jason99: It hasn't changed significantly, no. |
| 21:52.10 | Jason99 | jsmith: On 1.2 everything was fine.. but on 1.4 with nat=yes, using a D-Link DVG, when i receive a call waiting and press flash to answer it the two calls are conferenced like a 3-way call.. when nat=no, that problem does not occur.. |
| 21:52.47 | jsmith | Jason99: Very strange... can you get a SIP trace and post it to the bug tracker? |
| 21:52.57 | Jason99 | sure |
| 21:53.11 | Jason99 | where is the bug tracker? |
| 21:53.24 | Qwell | ~bugs |
| 21:53.54 | Jason99 | bot down? :P |
| 21:54.04 | Qwell | bugs.digium.com |
| 21:54.51 | jsmith | ~ping |
| 21:54.52 | jbot | pong |
| 21:55.47 | vap0rtranz | the book is awesome, but it should have said in the first sentence: ... everything's an extension. something just "clicked" b/c of that realization |
| 22:01.44 | BobLutz | Awesome Guys!!!!!! |
| 22:01.46 | *** part/#asterisk BobLutz (n=miles@d60-65-93-136.col.wideopenwest.com) |
| 22:02.51 | Katty | is there anything like hud, but better? |
| 22:02.58 | *** join/#asterisk mmlj4 (n=jkelly@ip70-171-94-246.no.no.cox.net) |
| 22:03.38 | fujin | haha |
| 22:06.29 | husimon | "You have 39 new messages" |
| 22:06.31 | husimon | WTF |
| 22:07.15 | husimon | 77777777777777777777777777777777777777777777777777 |
| 22:07.43 | husimon | 171717171717171717171717171 |
| 22:07.46 | husimon | is what I mean |
| 22:10.17 | *** join/#asterisk metfan2007 (n=metfan20@201.103.80.238) |
| 22:10.20 | metfan2007 | Hi all! |
| 22:10.41 | metfan2007 | Any idea why I get a discontinuous invite tone when using DISA? |
| 22:10.46 | husimon | what would cause a phone to call me over and over |
| 22:10.54 | husimon | and leave messages of silence |
| 22:11.07 | *** join/#asterisk VJFROMGT (n=vjfromgt@user-387g9sf.cable.mindspring.com) |
| 22:11.18 | VJFROMGT | when i do modprobe zaptel i get Error inserting zaptel (/lib/modules/2.6.9-34.0.2.ELsmp/extra/zaptel.ko): Unknown symbol in module |
| 22:11.56 | *** join/#asterisk Keltus (i=Keltus@about/cooking/nakedchef/beefstew/Keltus) |
| 22:11.57 | jsmith | VJFROMGT: Does your currently running kernel match the kernel headers that you have installed? |
| 22:13.10 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 22:13.11 | Katty | is there a way to barge in on both the extension and the person they're talking to and talk to both at the same time? |
| 22:13.14 | VJFROMGT | jsmith, i am not sure, i am a noob |
| 22:13.25 | Katty | rather than just whispering the extension |
| 22:13.26 | tzafrir | VJFROMGT, use './install_prereq test' in the zaptel directory |
| 22:13.27 | VJFROMGT | i isntalled zaptel via rpm |
| 22:14.29 | tzafrir | VJFROMGT, what error message do you actually get from the kernel? |
| 22:14.37 | tzafrir | dmesg | tail -n 2 |
| 22:15.02 | VJFROMGT | zaptel: Unknown symbol oslec_echo_can_update |
| 22:15.02 | VJFROMGT | zaptel: Unknown symbol oslec_echo_can_create |
| 22:15.08 | *** join/#asterisk Mikeonline (i=Mike@p57A7EFC7.dip.t-dialin.net) |
| 22:15.12 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 22:15.12 | Mikeonline | hi |
| 22:15.28 | tzafrir | VJFROMGT, you're missing the package oslec |
| 22:15.34 | tzafrir | or actually |
| 22:15.39 | VJFROMGT | ok,, ill isntall oslec |
| 22:15.42 | VJFROMGT | antyhing else? |
| 22:15.43 | tzafrir | the package with the kernel module oslec |
| 22:15.48 | tzafrir | what distro is that? |
| 22:15.54 | VJFROMGT | centos |
| 22:16.12 | tzafrir | what RPMs? where from? |
| 22:16.37 | VJFROMGT | rpm was pulled from a freebpx cd |
| 22:17.33 | tzafrir | VJFROMGT, what version of Zaptel is it? |
| 22:17.53 | tzafrir | Do you realize you use an ancient version of the Centos kernel? |
| 22:17.58 | VJFROMGT | zaptel-1.2.10-2.1567 |
| 22:18.15 | VJFROMGT | yes, there are other reasons why i need to use this centos |
| 22:18.16 | tzafrir | Do you actually have zaptel hardware? |
| 22:18.25 | VJFROMGT | no, i need ztdummy thouhg |
| 22:18.46 | lesouvage | Katty: afaik there is not a ready to go app but you could do this with some local channel magic and conferences I think. |
| 22:18.50 | *** join/#asterisk cleone (i=cleo@41.251.73.80) |
| 22:19.13 | Katty | lesouvage: yeah, maybe. |
| 22:19.19 | Katty | lesouvage: i can't quite figure out how to get there tho :/ |
| 22:19.47 | Katty | lesouvage: but you're right. something combining zapbarge and chanspy |
| 22:19.56 | Katty | lesouvage: and dumping it into a meetme conference |
| 22:19.56 | *** join/#asterisk marv[work] (n=timr@router.asteriasgi.com) |
| 22:20.18 | marv[work] | Hmm, I can't find the URL to the disclaimer. |
| 22:20.20 | Mikeonline | may i ask a very short question about variable evaluation? |
| 22:20.32 | russellb | marv[work]: s/disclaimer/license agreement/ |
| 22:20.34 | russellb | also, there is no URL |
| 22:20.58 | russellb | i think you get it when you click the "sign license" link when logged in to bugs.digium.com |
| 22:21.17 | Qwell | Mikeonline: if the question is an even number of words, and the third word ends in a v. |
| 22:21.47 | lesouvage | katty: if you join the two legs in a conference the third "leg" can join. |
| 22:22.02 | Qwell | Mikeonline: ask away - that's why we're here. |
| 22:22.20 | Mikeonline | oh i just searched for a word ending by v ;-) |
| 22:22.28 | Mikeonline | does anyone know why exten => h,1,DeadAGI(my.sh,$(FaxID)) is sent as $(FaxID) instead of the value of $(FaxID)? |
| 22:22.50 | Qwell | because $() doesn't evaluate |
| 22:22.52 | Qwell | ${} does |
| 22:23.15 | Mikeonline | argl |
| 22:23.31 | Mikeonline | its always the same after some times.. you get blind :( |
| 22:23.37 | Mikeonline | thx anyway |
| 22:24.03 | *** join/#asterisk Meaw (n=dino@a130-35.adsl.paltel.net) |
| 22:25.19 | Katty | lesouvage: do you know of a wiki page on how to dump two legs into a conference? |
| 22:25.50 | russellb | Redirect ftw |
| 22:25.56 | Katty | oh? |
| 22:26.04 | russellb | mhm |
| 22:26.22 | russellb | available as an app, and manager action |
| 22:26.39 | Katty | oh? |
| 22:26.46 | Katty | wiki? |
| 22:26.46 | Kobaz | i make a call from a sip/iax phone, and dial out onto the pstn. When connecting to certain destinations (especially cell phone voicemail), i will miss the first 5 seconds of the audio stream... anyone know how to fix this? |
| 22:28.19 | Kobaz | cleone: i think you missed some letters |
| 22:28.54 | Katty | russellb: http://www.voip-info.org/wiki/index.php?page=Asterisk+manager+Example%3A+Redirect <- something like that? |
| 22:29.06 | Kobaz | 14:31 <cleone> d u have iax? |
| 22:29.10 | Kobaz | i'm not sure what that means |
| 22:29.13 | cleone | kobaz : i mean iaxcomm* |
| 22:29.17 | cleone | kobaz : i mean iaxcomm |
| 22:29.31 | Kobaz | yeah i have it somewhere, it's kinda crappy |
| 22:29.48 | cleone | can u call me? |
| 22:29.58 | cleone | i want to test |
| 22:29.59 | Kobaz | umm |
| 22:30.04 | Kobaz | okay |
| 22:30.13 | russellb | Katty: probably |
| 22:30.15 | russellb | i don't know |
| 22:30.19 | russellb | i know nothing |
| 22:30.33 | Katty | :< |
| 22:30.37 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 22:30.55 | Kobaz | cleone: if you want me to call you, i'll need the hostname and a login |
| 22:31.17 | Katty | [TK]D-Fender: i figured out how to do it. sorta. http://www.voip-info.org/wiki/index.php?page=Asterisk+manager+Example%3A+Redirect |
| 22:31.33 | Katty | [TK]D-Fender: now's it's just taunting me |
| 22:31.37 | cleone | kobaz: i don't know how to use it |
| 22:32.03 | Kobaz | cleone: i think you should read up on some asterisk then |
| 22:32.19 | cleone | ok |
| 22:32.58 | Mikeonline | btw $FaxID is reserved for the Fax ID Number provided by chan_capi so you need to use sth like $myFaxID ;-) |
| 22:33.18 | Kobaz | cleone: once you have an iax account set up, you just set the hostname and user/pass and you're good to go |
| 22:33.45 | cleone | kobaz: d u have documentation |
| 22:33.53 | Kobaz | cleone: you're killing me |
| 22:34.27 | Kobaz | cleone: http://www.asteriskguru.com/ |
| 22:35.19 | Katty | does anyone have any redirect examples they can pastebin for me? |
| 22:36.41 | lesouvage | katty: you could fork the call and route both legs into a conference. |
| 22:37.31 | Katty | lesouvage: do you have a wiki? |
| 22:37.37 | Katty | lesouvage: any sample pastebin inflimation? |
| 22:38.53 | cleone | kobaz: thanks |
| 22:40.55 | lesouvage | katty: I have done this once. searching for it. forking by local channels is real asterisk magic. |
| 22:42.14 | *** part/#asterisk marv[work] (n=timr@router.asteriasgi.com) |
| 22:42.36 | Katty | i bet Corydon76-dig would know |
| 22:46.19 | Corydon76-dig | Eh? |
| 22:46.41 | Katty | Corydon76-dig: i bet you know how to get a SIP channel and a ZAP channel into a meetme. |
| 22:46.56 | Katty | Corydon76-dig: so a third party can 'barge' in...in a shiny automated fashion |
| 22:47.17 | Corydon76-dig | 1) Launch gastman |
| 22:47.27 | Corydon76-dig | 2) Create a conference icon |
| 22:47.36 | Corydon76-dig | 3) Drag both channels to the conference icon |
| 22:47.39 | Katty | :< |
| 22:47.53 | Corydon76-dig | You think I'm joking. |
| 22:48.02 | Corydon76-dig | That's how I do it |
| 22:48.18 | Katty | well, i don't think our sales staff will quite do that. |
| 22:48.28 | Corydon76-dig | except that gastman usually stays up all the time |
| 22:48.34 | Katty | they want to dial an extension, and put in the person they want to eavesdrop on |
| 22:48.43 | Katty | and be magically connected to both channels in a conference call |
| 22:49.15 | Corydon76-dig | Well, if not talking, you can use ChanSpy |
| 22:49.26 | Katty | Corydon76-dig: talking the both parties is the problem. |
| 22:49.40 | Katty | Corydon76-dig: i just saw your name all over app_ManagerRedirect and thought you might know (= |
| 22:49.56 | Corydon76-dig | newp |
| 22:49.58 | Katty | k |
| 22:50.11 | lesouvage | katty: I'm trying to get an example. it can be done. |
| 22:50.24 | Katty | lesouvage: well if ya find something, /query me (= |
| 22:50.34 | Katty | lesouvage: i'mma go home and try to forget all of this stuff for the night |
| 22:50.42 | Corydon76-dig | I do a lot of IVR, not so much black ops |
| 22:51.03 | Katty | Corydon76-dig: i'm convinced ChannelRedirect will do it. |
| 22:51.15 | Katty | Corydon76-dig: i just can't find an example to pick apart to see how it's done |
| 22:52.50 | Katty | oh well |
| 22:55.45 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 22:55.49 | fujin | Katty: can probably make use of app_conference to do it |
| 22:56.00 | fujin | make a button so that you can force a and b into a conference |
| 23:04.50 | *** join/#asterisk Isaiah (n=Spam@74-46-51-104.dr01.hnvr.mi.frontiernet.net) |
| 23:06.54 | Isaiah | What does asterisk dead but subsys locked mean? I get it when I type /etc/init.d/asterisk status |
| 23:07.35 | fujin | script to parse voicemail.conf and convert it into RealTime Voicemail: http://rafb.net/p/ABMMgr65.html |
| 23:07.42 | fujin | (basic, but it works ;P) |
| 23:08.19 | *** join/#asterisk anthm (n=anthm@mb80736d0.tmodns.net) |
| 23:08.19 | *** mode/#asterisk [+o anthm] by ChanServ |
| 23:11.28 | *** join/#asterisk Docfxit (n=none@cpe-72-129-118-135.socal.res.rr.com) |
| 23:11.56 | tzafrir | Isaiah, lock file still exists? |
| 23:12.11 | Isaiah | Yeah I does.. I think I found the problem through |
| 23:12.53 | Isaiah | Yeah, call fixed |
| 23:12.55 | Isaiah | all** |
| 23:13.08 | Isaiah | thanks for the help tzafrir anyway :) |
| 23:15.26 | *** part/#asterisk Isaiah (n=Spam@74-46-51-104.dr01.hnvr.mi.frontiernet.net) |
| 23:16.17 | VJFROMGT | after installing a new codec, how can i load them withouth restarting asterisk |
| 23:18.02 | fujin | module load codec_x.so |
| 23:24.03 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 23:24.26 | Mikeonline | is there a maximum number of DeadAGI parameters? |
| 23:24.40 | Corydon76-dig | 100, I think |
| 23:24.59 | VJFROMGT | my iax bandwidth is nowhere near to symetrical, its amlost 2:1 , can anyone help? |
| 23:25.10 | Mikeonline | hm after adding the fifth parameter for my script it doesnt work anymore |
| 23:25.10 | VJFROMGT | i already have ztdummy installed |
| 23:28.53 | [TK]D-Fender | VJFROMGT, Whats to help? Your connection is what it is. Whats the problem? |
| 23:29.02 | Mikeonline | maybe the paramter has problems with the @ sign in the mail address |
| 23:29.12 | VJFROMGT | tk chan_iax2.c: Unable to support trunking on peer 'Sat_out' without zaptel timing |
| 23:29.29 | VJFROMGT | how can i sove that issue? |
| 23:29.45 | VJFROMGT | i usually get it resovled by making sure ztdummy is loaded |
| 23:30.06 | russellb | compile and install zaptel before compiling asterisk |
| 23:30.12 | russellb | and then load ztdummy before running asterisk |
| 23:31.12 | VJFROMGT | since my zaptel works, does that mean it is just the sequence of loading or should i remove zaptel and ansterisk and start from scratch? |
| 23:32.14 | [TK]D-Fender | VJFROMGT, looks like wven with ztdummy loaded you have no recompiled * with zaptel support. Please do so. |
| 23:32.35 | VJFROMGT | thanks tk |
| 23:32.39 | VJFROMGT | ill remvoe and recompile |
| 23:32.44 | vap0rtranz | why would From: and To: in sip headers be identical?? |
| 23:33.02 | [TK]D-Fender | vap0rtranz, Because the voices need company. |
| 23:33.17 | vap0rtranz | [TK]D-Fender: hehe |
| 23:33.49 | [TK]D-Fender | vap0rtranz, And they'd be the same if a SIP user account was dialing an exten on the local * box that happens to be numbered/named identically. |
| 23:34.07 | vap0rtranz | it looks like there's some secret changes made b/w 1.2 -> 1.4, because i haven't been able to simply port the registrations over |
| 23:34.20 | adeel | which one is better, hylafax or hylafax+ ? |
| 23:34.34 | [TK]D-Fender | vap0rtranz, No there are a lof of documented changes going from 1.2 to 1.4 |
| 23:35.07 | vap0rtranz | [TK]D-Fender: so i should throw away the book! |
| 23:37.19 | flush | yo damnit |
| 23:37.22 | flush | mayday |
| 23:37.52 | *** join/#asterisk quigon (n=matias@200.61.187.185) |
| 23:37.52 | flush | ive read http://downloads.oreilly.com/books/9780596510480.pdf till they start configuring voip cause i dont have any provider |
| 23:38.12 | flush | i got my phone working with the example, i call extension 500 and i get the repeat thing.. |
| 23:38.25 | flush | but when i call here with my skype account, asterisk doesnt ring the phone |
| 23:38.33 | flush | any idea please im sad |
| 23:39.04 | flush | and when i try to compose something on the phone plugged in the FXS module |
| 23:39.05 | vap0rtranz | flush: cli? |
| 23:39.11 | flush | i get the fast busy tone |
| 23:39.14 | flush | what you mean |
| 23:39.58 | *** join/#asterisk quigon (n=matias@200.61.187.185) |
| 23:41.39 | vap0rtranz | flush: what do you see from watching #asterisk -r |
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